| 00:00.05 | blitzrage | EmleyMoor: ok thats what I thought. Just grab a logo and set the print size |
| 00:00.21 | EmleyMoor | I will have a go |
| 00:00.36 | EmleyMoor | The label I currently have has a Sunderland number on it |
| 00:01.42 | *** join/#asterisk ccesario (n=ccesario@matriz.isic.com.br) |
| 00:07.33 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 00:08.34 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 00:11.57 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 00:14.59 | *** join/#asterisk Kyler (n=chatzill@74.132.200.229) |
| 00:17.05 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 00:26.38 | *** join/#asterisk ccesario (n=ccesario@matriz.isic.com.br) |
| 00:28.27 | *** join/#asterisk yassine (n=yassine@xdsl-84-44-154-91.netcologne.de) |
| 00:39.54 | *** join/#asterisk BSDTech (n=RNeese@adsl-69-230-170-5.dsl.irvnca.pacbell.net) |
| 00:55.53 | backblue_ | robin_z: increase the value on the last line of the misdn.conf |
| 00:56.18 | backblue_ | offcourse it suports callerid, it's standart in misdn, should work by default. |
| 00:57.36 | *** join/#asterisk andresmujica (n=AndresMu@201.245.232.137) |
| 01:00.21 | *** join/#asterisk X-Rob_ (n=Rob@dsl-202-173-151-24.qld.westnet.com.au) |
| 01:16.19 | *** join/#asterisk Flosoft (n=admin@213.219.143.140.adsl.dyn.edpnet.net) |
| 01:16.41 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool1-11.mtco.com) |
| 01:27.56 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-42.ph.ph.cox.net) |
| 01:28.44 | *** join/#asterisk uwe (n=uwe@213.6.12.200) |
| 01:29.59 | *** join/#asterisk stuq_ (n=stuq@user-12lcqia.cable.mindspring.com) |
| 01:31.20 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 01:31.54 | *** join/#asterisk dasenjo (n=dasenjo@63.245.86.186) |
| 01:32.02 | *** join/#asterisk Grubs (n=Miranda@c220-239-223-90.eburwd3.vic.optusnet.com.au) |
| 01:32.56 | Grubs | I see so many people installing asterisk on multi-function boxes. Has asterisk been improved to the point where the analog encoding/decoding is more robust to interrupts? |
| 01:33.07 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:40.09 | *** join/#asterisk bradoaks (n=bradoaks@dog.lost-habit.com) |
| 01:41.32 | *** part/#asterisk andresmujica (n=AndresMu@201.245.232.137) |
| 01:44.15 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 01:45.32 | *** join/#asterisk Corydon-w (n=tilghman@vcchgate.vcch01.springfield.tn.us.vcch.net) |
| 01:45.32 | *** mode/#asterisk [+o Corydon-w] by ChanServ |
| 01:46.58 | *** join/#asterisk tengulre (n=tengulre@221.11.5.182) |
| 01:48.06 | *** join/#asterisk mhnoyes_ (n=mhnoyes@dialup-4.246.239.252.Dial1.SanJose1.Level3.net) |
| 01:52.21 | robin_sz | Grubs, not really, no. its best not to put too omuch "other stuff" on the box all the same |
| 01:53.23 | robin_sz | backblue_, ok , ill try increasing that value ... I guess its possible my poxy provider doesnt send CID on the circuit .. its BT, they are a bit crap at times. |
| 01:53.34 | [TK]D-Fender | Grubs : depends how much you're asking of it. |
| 01:53.55 | robin_sz | true |
| 01:54.05 | robin_sz | sip to sip seems to work well on any sort of box |
| 01:54.22 | robin_sz | its when you start adding PCI cards for ISDN etc that the troubles start |
| 01:55.32 | orlock | Is anybody here using a multi-number sip account? |
| 01:56.07 | backblue_ | robin_sz: pickup a isdn phone, and test it. |
| 01:56.39 | robin_sz | not a bad plan |
| 01:57.01 | *** join/#asterisk MoutaPT (n=root@a213-22-40-63.cpe.netcabo.pt) |
| 01:57.02 | robin_sz | I might get an ISDN phone, just to have a back-up should my * box shit itself one day |
| 01:57.22 | backblue_ | robin_sz: sorry, you have to decrease the value in misnd.conf, not incriese! |
| 01:57.57 | *** join/#asterisk Dibbler (n=Dibbler@dsl-217-155-254-174.zen.co.uk) |
| 01:58.36 | robin_sz | k |
| 01:58.55 | robin_sz | I'll save that fun for later in the week |
| 02:00.51 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 02:00.51 | *** join/#asterisk gtkspert (n=gtkspert@gateless.info) |
| 02:01.19 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 02:04.20 | orlock | Can anybody tell me if asterisk uses the sip Invite header or the sip To: header when determining DID's |
| 02:06.33 | orlock | Anybody? |
| 02:07.03 | *** join/#asterisk apardo (n=apardo@87.217.147.43) |
| 02:07.39 | MoutaPT | it's the To: header |
| 02:07.42 | MoutaPT | orlock |
| 02:07.49 | orlock | hmm |
| 02:07.55 | orlock | thats how it should be |
| 02:08.10 | MoutaPT | if you have more than one DID |
| 02:08.13 | MoutaPT | from your telco |
| 02:08.13 | orlock | its FreePBX though, so nobody here will want to help with that :) |
| 02:08.22 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 02:08.28 | MoutaPT | you should look a fucntion in dialplan |
| 02:08.31 | MoutaPT | named |
| 02:08.32 | orlock | MoutaPT: yeah, one sip account with 10 DID's assigned |
| 02:09.22 | MoutaPT | <PROTECTED> |
| 02:09.22 | MoutaPT | Sets a channel variable to the content of a SIP header |
| 02:09.30 | MoutaPT | check this over the wiki |
| 02:09.34 | orlock | MoutaPT: ahh, cool |
| 02:09.34 | orlock | yeah |
| 02:09.39 | MoutaPT | or test it in your dialplan |
| 02:09.40 | *** join/#asterisk Az_au (n=az@216.127.73.119) |
| 02:09.41 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 02:09.43 | orlock | i thought it would be something like that |
| 02:09.50 | MoutaPT | you are looking for variable |
| 02:09.51 | MoutaPT | to |
| 02:09.53 | MoutaPT | ;) |
| 02:09.56 | MoutaPT | good luck |
| 02:11.43 | *** join/#asterisk malcolmd (i=malcolmd@nat/digium/x-5427194079719780) |
| 02:14.03 | orlock | MoutaPT: thanks dude, it will take some work, but i know i am not nuts now :) |
| 02:14.35 | MoutaPT | ;) |
| 02:15.53 | *** join/#asterisk Dibbler (n=Dibbler@dsl-217-155-254-174.zen.co.uk) |
| 02:16.55 | *** join/#asterisk asberger (n=nb@cpe-76-170-93-79.socal.res.rr.com) |
| 02:18.31 | *** join/#asterisk leoncamel (n=leoncame@219.238.107.107) |
| 02:28.48 | *** join/#asterisk Dibbler (n=Dibbler@dsl-217-155-254-174.zen.co.uk) |
| 02:29.42 | robin_sz | backblue_, this look correct? |
| 02:29.44 | robin_sz | /sbin/modprobe --ignore-install hfcpci protocol=0x2 layermask=0xf |
| 02:29.44 | robin_sz | modprobe mISDN_dsp debug=0x0 options=0 dtmftreshold=30 |
| 02:31.11 | *** join/#asterisk jhnjwng (n=wj1918@pool-141-150-92-7.nwrk.east.verizon.net) |
| 02:31.20 | backblue_ | robin_sz: dunno, tired |
| 02:31.22 | backblue_ | cya |
| 02:32.48 | *** join/#asterisk bkw__ (n=brian@24.249.5.154) |
| 02:33.39 | robin_sz | coo, dtmf detection does work! |
| 02:36.21 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 02:36.26 | *** part/#asterisk MoutaPT (n=root@a213-22-40-63.cpe.netcabo.pt) |
| 02:38.02 | Grubs | lol - fell I asleep! Back to asterisk on multi-function boxes - if just using sip/iax without any PCI cards I can see why it wouldnt matter what else the box was doing. |
| 02:38.58 | Grubs | When I set up asterisk over a year ago everyone here at the time was adament I'd need a dedicated box even with a single FXO card (TDM). |
| 02:39.31 | robin_sz | probably true today |
| 02:39.52 | robin_sz | interrupt handling is the killer |
| 02:40.34 | robin_sz | zttest is your friend here |
| 02:42.05 | Grubs | I know zttest can tell you if you are sharing an interrupt - does it do more (googling now) |
| 02:42.52 | *** join/#asterisk Dibbler (n=Dibbler@dsl-217-155-254-174.zen.co.uk) |
| 02:46.56 | robin_sz | it shows you if interrupts are being handled in a timely fashion ... or more to the point, if they are not |
| 02:47.02 | robin_sz | for whatever reason |
| 02:47.14 | robin_sz | CPU buy ness, sharedness, crapness |
| 02:47.23 | robin_sz | busy-ness |
| 02:48.56 | *** join/#asterisk mhnoyes__ (n=mhnoyes@dialup-4.246.18.17.Dial1.SanJose1.Level3.net) |
| 02:49.47 | Grubs | I see. zttest reports all 99.987793% on my clean dedicated box. I might install to my ClarkConnect web proxy/gate way and see if it drops (uses the same hardware) |
| 02:50.55 | robin_sz | thats the exact same value I get ... |
| 02:53.07 | *** join/#asterisk santiago (n=santiago@208.195.214.146) |
| 03:02.15 | *** join/#asterisk Ryanw (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 03:03.40 | Ryanw | whats the best free soft phone with multiple sip accounts? |
| 03:04.27 | robin_sz | hmm ... i only want cdr_mysql ... how do I turn off the custom/CSV stuff? |
| 03:07.16 | Ryanw | you could always symlink the csv to /dev/null if ya can't find out where to turn it off. |
| 03:07.33 | robin_sz | hmmm |
| 03:09.04 | *** join/#asterisk Anson (n=anson@va-charlottesville-cdnt1-bg8-11a-b-170.chvlva.adelphia.net) |
| 03:09.11 | Qwell | uhh |
| 03:09.14 | Qwell | or just unload the module |
| 03:10.30 | robin_sz | ahh, in modules.conf? |
| 03:11.58 | *** join/#asterisk docelmo (n=vircuser@c-68-32-143-73.hsd1.de.comcast.net) |
| 03:21.55 | *** join/#asterisk Mattwj2005 (n=Matt@c-76-17-131-68.hsd1.mn.comcast.net) |
| 03:22.10 | Mattwj2005 | hey guys :) |
| 03:22.44 | Mattwj2005 | I am going to rebuild my asterisk server tonight |
| 03:22.47 | Mattwj2005 | gentoo :) |
| 03:24.45 | *** part/#asterisk Ryanw (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 03:24.52 | Mattwj2005 | anyone here? |
| 03:24.59 | bradoaks | i may be being very dense here, but i've installed asterisk-now appliance from rpath and cannot login with the admin password i set using the passwd command as root for the admin user. I don't remember setting an application-level password during install. any ideas? i may just reinstall. |
| 03:25.08 | bradoaks | hi Mattwj2005. |
| 03:25.25 | Mattwj2005 | hey bradoaks :) |
| 03:26.07 | bradoaks | Mattwj2005: there have been a handful of comments over the past hour and a handful in the hour before that. |
| 03:26.17 | bradoaks | so folks are around, but not chatty. |
| 03:26.21 | Mattwj2005 | lol |
| 03:26.24 | Mattwj2005 | got you.... |
| 03:26.43 | Mattwj2005 | this is the start of the weekend for me......for most people it is the end of the weekend |
| 03:28.05 | Mattwj2005 | yeah I have a small personal PBX at my house....not real special...just for cheap long distance :) |
| 03:28.32 | Grubs | monday afternoon here |
| 03:28.53 | Mattwj2005 | where are you at Grubs? I am in Minneapolis |
| 03:29.02 | Grubs | Melbourne - Australia |
| 03:29.26 | Grubs | just upgrading my home/office pbx |
| 03:29.42 | Mattwj2005 | nice....how many users? |
| 03:29.56 | Grubs | only 3 extensions. |
| 03:31.03 | Grubs | using an old 1U poweredge server - PIII 550. Works like a charm and virtually silent after I removed most of the fans |
| 03:32.24 | *** join/#asterisk Dibbler (n=Dibbler@dsl-217-155-254-174.zen.co.uk) |
| 03:38.49 | *** join/#asterisk Flosoft (n=admin@213.219.143.140.adsl.dyn.edpnet.net) |
| 03:41.51 | JT | Grubs: removed most of the fans... from a 1RU server |
| 03:41.53 | JT | :o |
| 03:41.54 | JT | ... |
| 03:46.26 | Grubs | its OK - CPU is only PIII 550 and I replaced the passive heatsink with a low-profile unit with a slow fan. This allowed me to remove 4 of the screaming 40mm fans. PSU is cooled using a PCI-slot blower (high static pressure) that occupies one drive slot. |
| 03:47.17 | Grubs | Server only uses 36W of power and is nice and quiet. |
| 03:47.23 | JT | hrm |
| 03:47.32 | JT | redundant power supply? |
| 03:47.45 | Grubs | no |
| 03:47.56 | JT | they're quite rare in 1RU |
| 03:49.31 | *** join/#asterisk NDT (n=noone@cpe-74-70-211-81.nycap.res.rr.com) |
| 03:49.34 | Grubs | Poweredge 350 is a toy compared to modern servers. but perfect for asterisk. I have a second one as a ClarkConnect gateway/squid proxy. |
| 03:51.38 | JT | heh, i only tend to put asterisk on servers with redundant power supplies and RAID1 drives |
| 03:53.04 | *** join/#asterisk bmg505 (n=leon@c1-136-3.rndf.isadsl.co.za) |
| 03:53.08 | Grubs | fine - but mine serves a home and home-office with 3 extensions - so its not a prob. Our voip provider offers failover to mobiles automatically if our server is offline. |
| 03:56.23 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 03:58.15 | *** join/#asterisk Mattwj2005 (n=Matt@c-76-17-131-68.hsd1.mn.comcast.net) |
| 03:58.53 | Mattwj2005 | so what distros do you guys use? |
| 03:59.39 | Grubs | Debian Sarge 3.1 - minimal install (CLI only). |
| 04:00.02 | Mattwj2005 | I have used that too |
| 04:00.13 | Mattwj2005 | I use Gentoo 2006.1 CLI too |
| 04:03.21 | JT | Grubs: fair enough, i was talking about my home servers too :) |
| 04:03.25 | JT | what voip provider is that? |
| 04:03.57 | *** join/#asterisk Dibbler (n=Dibbler@dsl-217-155-254-174.zen.co.uk) |
| 04:04.02 | Grubs | Nehos.net (based in Brisbane - AU) |
| 04:04.19 | JT | hrm never heard of them |
| 04:04.22 | JT | will look into it |
| 04:05.02 | Grubs | not the cheapest by a long shot. But trouble free. You get what you pay for. |
| 04:05.21 | JT | their page still hasn't loaded :P |
| 04:05.32 | JT | just got a timeout |
| 04:06.16 | Grubs | instant load here |
| 04:06.51 | JT | oh, www.nehos.net |
| 04:06.56 | JT | their non www one is stuffed |
| 04:06.59 | Grubs | ah |
| 04:08.31 | *** join/#asterisk dsd_ (n=dsd@gentoo/developer/dsd) |
| 04:11.34 | *** part/#asterisk dsd_ (n=dsd@gentoo/developer/dsd) |
| 04:13.17 | *** join/#asterisk Dibbler (n=Dibbler@dsl-217-155-254-174.zen.co.uk) |
| 04:14.07 | Mattwj2005 | with Gentoo you can run the installer in ssh :) |
| 04:15.57 | Sedorox | if anyone cares.. I just posted information on voip-info on how to get a Vina integrator/Lucent Connectreach/many other names... working with Asterisk through a Cisco 1760 router |
| 04:16.49 | *** join/#asterisk Jeff81 (n=services@pool-72-71-240-199.cncdnh.fios.verizon.net) |
| 04:16.54 | Sedorox | I use Gentoo too btw |
| 04:17.10 | Mattwj2005 | oh yeah? |
| 04:17.29 | Mattwj2005 | what do you think of it sedorox? |
| 04:17.46 | Sedorox | I like Gentoo |
| 04:17.50 | tlp | Any of you guys use ASTLinux? |
| 04:17.57 | Mattwj2005 | do you do just a emerge asterisk? |
| 04:18.07 | Jeff81 | Hi, got a question that probablysounds stupid. I've been reading something about chan_modem.so that used to be included in asterisk but no longer is. Does anyone have this file and any other related files that I may be able to play around with? |
| 04:18.11 | tlp | I'm considering Debian for my production system. Currently doing testing on FreeBSD, but as it's not supported, gonna hop over to Linux. |
| 04:18.20 | Sedorox | yes.. and make sure you have the use flags right if you plan on using pri/zaptel/etc... |
| 04:19.21 | *** join/#asterisk insomnia41 (n=dasver96@c-69-254-230-163.hsd1.ks.comcast.net) |
| 04:20.06 | Mattwj2005 | oh okay |
| 04:20.17 | Mattwj2005 | my pbx is very basic :) |
| 04:20.41 | file | Jeff81: why do you ask? :D |
| 04:20.50 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001346f7843f.vc.shawcable.net) |
| 04:21.05 | *** join/#asterisk jeffgus (n=jeffgus@marlene.zimage.com) |
| 04:22.35 | Jeff81 | wantedto play around with making asterisk work with a voice modem. I havea few old voice modems, and I've got a Skype account with a phone adapter. If I could get a voice modem working on Asterisk, it would be a simple way to connect to Skype. |
| 04:22.58 | file | the channel driver you are talking about was very very old, and only did half duplex on a few modems |
| 04:23.49 | Jeff81 | I read that in a forum, just figured I'd see if it worked with any of my modems. But I can't even find a copy of it. |
| 04:24.13 | file | it was removed because it was old and unused and nobody maintained it |
| 04:24.16 | Jeff81 | I've tried the uplink program to link Skype to SIP, but I can't seem to get that working. |
| 04:27.40 | *** part/#asterisk BSDTech (n=RNeese@adsl-69-230-170-5.dsl.irvnca.pacbell.net) |
| 04:27.55 | Jeff81 | do you know where I might be able to find these old drivers? Or something to make Skype connect to Asterisk? I have found a few different articles on connecting with the uplink program, but I'm doing something wrong and don't know what. Once I got an Asterisk extension to dial out on Skype, but had no audio (send or receive) and couldn't get incoming calls. |
| 04:28.50 | file | the old drivers are not the way to go... but if you really want them, you can probably grab them from an old SVN revision - and as for connecting to Skype I have no experience |
| 04:30.13 | Jeff81 | I apologize, I am not very experienced with all of this myself. Do you know where I could find these old revisions/ |
| 04:41.30 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 04:41.39 | *** join/#asterisk Gr1ncheux (n=devine@AStDenis-105-1-34-221.w80-8.abo.wanadoo.fr) |
| 04:42.12 | *** join/#asterisk robin_sz (n=robin@rapid2.gotadsl.co.uk) |
| 04:45.54 | *** join/#asterisk S2AnGeL (n=S2AnGeL@CPE000f66bd2bf3-CM000039529869.cpe.net.cable.rogers.com) |
| 04:46.36 | *** part/#asterisk S2AnGeL (n=S2AnGeL@CPE000f66bd2bf3-CM000039529869.cpe.net.cable.rogers.com) |
| 04:47.09 | *** part/#asterisk Mattwj2005 (n=Matt@c-76-17-131-68.hsd1.mn.comcast.net) |
| 04:56.45 | *** join/#asterisk [hC] (n=hardcore@S010600404521309c.vc.shawcable.net) |
| 04:56.54 | [hC] | anyone played around with the new polycom 2.x firmware yet? |
| 04:59.43 | *** join/#asterisk olinux (i=olinux@ip68-107-4-202.sd.sd.cox.net) |
| 05:00.57 | *** join/#asterisk arcanine (n=arcanine@203.82.44.179) |
| 05:06.58 | *** join/#asterisk JT_ (n=jon@unaffiliated/jt) |
| 05:09.36 | [TK]D-Fender | [hC] : Yup, seems fine |
| 05:11.50 | Supaplex | it pooped on the carpet! bad firmware! ;) |
| 05:13.53 | Supaplex | sup file :) |
| 05:15.16 | file | I am... working on the bug tracker early, taking a break from a project |
| 05:15.17 | file | you? |
| 05:16.04 | Supaplex | thinking of a 7-8hr nap |
| 05:16.14 | file | good plan |
| 05:17.47 | *** join/#asterisk apardo (n=apardo@87.217.147.43) |
| 05:18.18 | Sedorox | ok.. you know your a voip geek when you girlfriend says she'll call you.. and you ask on which phone... |
| 05:19.31 | Supaplex | home|cell|work |
| 05:21.34 | Sedorox | well I was just talking about all the phones I have setup in the lab.. so it was actually ment as a joke... but just sounds funny |
| 05:21.44 | Sedorox | I have, with me right now, a cell, voip, and 1 analog |
| 05:21.46 | Sedorox | 2 analog |
| 05:22.48 | Supaplex | and I like turning statements like that on its head :) |
| 05:23.12 | Sedorox | :p |
| 05:23.26 | [TK]D-Fender | Sedorox : I've got 3 Polycom phones in my bedroom at home, an ATA that isn't plugged ATM, and am connected to about 1.2 dozen servers worldwide :) |
| 05:23.38 | Sedorox | heh |
| 05:23.40 | Sedorox | nice |
| 05:23.45 | [TK]D-Fender | Sedorox : But you've got me on the "analog line" bit... won't see that here :D |
| 05:23.50 | Sedorox | hehe |
| 05:24.02 | [TK]D-Fender | 3 cheers for Dry-line DSL! |
| 05:24.16 | Sedorox | I have a grandstream, a 7912 (which I can't use right now.. think I need CCM for it...) and 2 7960's |
| 05:24.26 | Sedorox | then I have a Lucent ConnectReach channel bank |
| 05:24.32 | Sedorox | which is how the analogs are connected |
| 05:24.39 | Sedorox | along with a fxs vic in the cisco router |
| 05:24.49 | file | Polycom IP600 and a PAP2-NA! yay |
| 05:24.55 | Sedorox | hehe |
| 05:25.03 | Sedorox | I wanna pick up a polycom still |
| 05:25.14 | Sedorox | 'cept I just had to get a car.. so thats pushed off a bit |
| 05:26.55 | [TK]D-Fender | Sedorox : Well I was a little gung-ho on my purchases and Am looking to sell off 2 and replace with something higher end perhaps. |
| 05:27.14 | [TK]D-Fender | Sedorox : But they are great as test phones for business. |
| 05:27.27 | Sedorox | the poly's? |
| 05:28.24 | [TK]D-Fender | Sedorox : yup |
| 05:28.34 | Sedorox | what model.. and how much you asking? :p |
| 05:28.40 | [TK]D-Fender | Sedorox : Then again.... what do I need more phones for... |
| 05:28.43 | file | two trillion dollars! |
| 05:28.49 | [TK]D-Fender | Sedorox : I have a 301, 430, and 501 |
| 05:28.53 | file | [TK]D-Fender: it's not like you get phone calls |
| 05:29.03 | [TK]D-Fender | file : I do actually. |
| 05:29.11 | file | [TK]D-Fender: blasphemy! |
| 05:29.24 | Sedorox | lol |
| 05:30.03 | [TK]D-Fender | file : Every other week! |
| 05:30.04 | Sedorox | I'd love the 501... but I probably won't even be able to afford the 301 :/ |
| 05:33.46 | [TK]D-Fender | file : It should, though i've registered a permanent domain as well now :) |
| 05:34.21 | file | yay permanent domain |
| 05:34.46 | [TK]D-Fender | file : Yup, I went and paid my dues to the Registrars From Hell :D |
| 05:36.59 | *** join/#asterisk Dibbler (n=Dibbler@dsl-217-155-254-174.zen.co.uk) |
| 05:43.59 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 05:44.02 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:50.39 | *** part/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 05:52.41 | Igbothom_III | ooh, aah, Glenn McGrath |
| 05:53.12 | *** join/#asterisk robin_sz (n=robin@rapid2.gotadsl.co.uk) |
| 05:59.22 | *** join/#asterisk [B]abyCool16 (n=java@server.ivinskis.kursenai.lm.lt) |
| 06:07.09 | *** join/#asterisk nb_ (n=nb@cpe-76-170-93-79.socal.res.rr.com) |
| 06:09.16 | *** join/#asterisk Un1x (i=Un1x@CPE000c419d026c-CM001225402bae.cpe.net.cable.rogers.com) |
| 06:16.45 | shellshark | anyone here a Freemason? |
| 06:17.50 | dlynes_laptop | You might want to try #freemasons |
| 06:17.59 | shellshark | ooo |
| 06:18.10 | shellshark | no dice ;) |
| 06:18.56 | dlynes_laptop | how about #rosachristian? |
| 06:19.38 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 06:20.09 | shellshark | what's that? |
| 06:20.22 | dlynes_laptop | It's another weird thing like freemasons :) |
| 06:20.37 | shellshark | masons are weird? |
| 06:20.49 | F | ... |
| 06:20.59 | dlynes_laptop | to me they are |
| 06:21.02 | dlynes_laptop | maybe not to everyone |
| 06:21.10 | shellshark | why's that? |
| 06:21.24 | dlynes_laptop | Just the whole secrecy thing |
| 06:21.41 | dlynes_laptop | and the eye of isis fascination or whatever it is |
| 06:21.45 | shellshark | the only thing that are really secret are the handshakes and passwords :) |
| 06:21.58 | shellshark | the all seeing eye... |
| 06:22.28 | dlynes_laptop | the meetings are all private, too |
| 06:22.45 | shellshark | that's their symbol for the "supreme being" above (they label it that way to be politically correct) |
| 06:22.52 | dlynes_laptop | and from what i understand there's a lot of really sinister requirements after you get to 14th level or something like that |
| 06:23.15 | shellshark | there are only 3 degrees of the standard freemasons |
| 06:23.26 | shellshark | apprentice, fellowcraft, and master |
| 06:23.52 | shellshark | there are other masonic-related organizations that have higher numbers, but that's different |
| 06:24.52 | shellshark | such as the scottish rite, having degrees 4-32, and the york rite having 4-14 (iirc.. york might have more than that) |
| 06:25.20 | dlynes_laptop | this was some masonic lodge in England |
| 06:25.50 | *** join/#asterisk Dibbler (n=Dibbler@dsl-217-155-254-174.zen.co.uk) |
| 06:26.09 | orlock | freemasons run the country! |
| 06:26.25 | dlynes_laptop | which country? |
| 06:26.32 | dlynes_laptop | USA? |
| 06:26.38 | orlock | yeah |
| 06:26.44 | orlock | according to the simpsons |
| 06:26.47 | shellshark | dlynes_laptop: there are MANY masonic lodges in england ;) |
| 06:26.50 | dlynes_laptop | Yeah...just look at your dollar bill |
| 06:26.50 | shellshark | orlock: hehehe |
| 06:27.01 | orlock | my grandfather was a mason |
| 06:27.03 | dlynes_laptop | It's got masonic symbols plastered all over it |
| 06:27.14 | orlock | its a conspiracy! |
| 06:27.18 | orlock | 23 skidoo! |
| 06:27.23 | shellshark | dlynes_laptop: yeah, but the people who founded the country were essentially all masons, and needed something to put on the currency ;) |
| 06:27.48 | dlynes_laptop | Yeah...from what I hear, every president except one has been a freemason, too |
| 06:27.52 | *** join/#asterisk insomnia41 (n=dasver96@c-69-254-230-163.hsd1.ks.comcast.net) |
| 06:27.56 | shellshark | not every |
| 06:27.59 | dlynes_laptop | I think the only exception was Clinton wasn't it? |
| 06:28.07 | shellshark | bush sr, bush jr, clinton were not |
| 06:28.18 | shellshark | lincoln was not |
| 06:28.34 | shellshark | there were a lot more that actually were not masons too |
| 06:28.55 | dlynes_laptop | Maybe it was all but Clinton that were members of the NRA, then |
| 06:29.07 | dlynes_laptop | NRA, masons, ... same crap different pile :) |
| 06:30.36 | *** join/#asterisk hmmhesays (n=Neg@24-117-135-28.cpe.cableone.net) |
| 06:32.10 | shellshark | ouch ;) |
| 06:32.28 | shellshark | why do you feel so negative about masons if you don't know anything about them? |
| 06:33.41 | *** join/#asterisk shtoom (n=godson@202-149-32-16.static.exatt.net) |
| 06:35.35 | *** join/#asterisk VibroMax (n=phisto@83.229.70.154) |
| 06:37.10 | dlynes_laptop | Knowing which presidents were masons and which ones weren't means knowing something about freemasons? |
| 06:37.38 | dlynes_laptop | I've got one friend that used to be a mason, and another that used to be a Rosa Christian...they both told me about masons |
| 06:41.30 | *** join/#asterisk aadilismail (n=aadilism@41-247-154-202.wol.net.pk) |
| 06:41.36 | aadilismail | hi |
| 06:46.59 | *** join/#asterisk Tebi_ (n=rantis@gw.aller.fi) |
| 06:51.02 | *** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net) |
| 06:52.19 | *** join/#asterisk sumasuma (n=kurukko@61.14.86.23) |
| 06:58.22 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 07:01.29 | *** join/#asterisk apardo (n=apardo@87.217.147.43) |
| 07:02.38 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:05.04 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net) |
| 07:08.09 | *** join/#asterisk jay (n=jay@lindalane.com) |
| 07:12.15 | *** join/#asterisk tr2x (n=alvar@80-218-150-90.dclient.hispeed.ch) |
| 07:13.28 | *** join/#asterisk tegioz (n=tegioz@62.87.59.222) |
| 07:19.47 | shellshark | dlynes_laptop: what did they tell you? |
| 07:21.12 | dlynes_laptop | Just a few things about all the weird stuff that happened at the meetings and how the levels get progressively more and more sinister |
| 07:21.36 | dlynes_laptop | Until finally after you've been in there for a while, they get you do more criminal like things |
| 07:21.45 | Sedorox | hmm |
| 07:21.55 | shellshark | wow, that's insanely misleading |
| 07:22.27 | dlynes_laptop | Well, i'm sure there's gotta be a reason why he left, and why he's still having nightmares about it |
| 07:22.33 | shellshark | the most criminal thing they've gotten me to do has been panhandling for the Salvation Army to raise funds for disadvantage children during christmas |
| 07:23.08 | dlynes_laptop | what level are you, though? |
| 07:23.21 | shellshark | i'm the highest degree a mason can be |
| 07:23.27 | shellshark | master mason |
| 07:24.08 | dlynes_laptop | odd |
| 07:24.20 | dlynes_laptop | I've heard these kinda things from other people, too |
| 07:24.31 | dlynes_laptop | Just didn't hear it in as much detail as I did from this fellow |
| 07:24.55 | shellshark | they do a lot of philantropy, fundraisers for good causes, public dinners, etc... i guess in some places of the world that could be considered criminal |
| 07:25.21 | dlynes_laptop | nah...there's other things they would get him to do such as laying down in a coffin for some reason or another, too |
| 07:25.35 | shellshark | whoa! |
| 07:25.42 | *** join/#asterisk JT (n=jon@CPE-141-168-62-0.nsw.bigpond.net.au) |
| 07:25.44 | shellshark | that's not right |
| 07:25.53 | shellshark | where was this at? |
| 07:26.04 | dlynes_laptop | That was at his 11th or 12th level |
| 07:26.14 | dlynes_laptop | It was his initiation rite for that level |
| 07:26.29 | shellshark | s/rite/ritual/ |
| 07:26.42 | shellshark | was that scottish rite or york rite? |
| 07:26.46 | dlynes_laptop | No idea |
| 07:26.56 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 07:26.56 | shellshark | ask him next time you talk to him, if you would |
| 07:27.04 | dlynes_laptop | I would imagine if it's either one of those, it'd be york as he's from England, not Scotland |
| 07:27.04 | shellshark | i'd be interested to find out |
| 07:27.20 | shellshark | they are both worldwide organizations |
| 07:27.28 | dlynes_laptop | ah |
| 07:27.31 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
| 07:27.47 | dlynes_laptop | anyways...the whole thing seemed pretty creepy |
| 07:28.09 | Aurs | is there a way to "kill" a sip channel that is hung up, but is still showing in "show channels" ? |
| 07:28.12 | dlynes_laptop | the former rosa christian is his wife |
| 07:28.21 | dlynes_laptop | She's got some weird tales about that organization, too |
| 07:28.35 | shellshark | well I can't speak on behalf of either the york rite or scottish rite, as I'm not a member of either, but the fundamental masonic organization (aka Blue Lodge) is nothing like that |
| 07:28.35 | dlynes_laptop | soft hangup sip/101 |
| 07:28.41 | dlynes_laptop | Aurs: or something similar |
| 07:29.14 | Aurs | k, thanks. will check that out |
| 07:29.57 | Aurs | have a call that have been active since friday.. hehe |
| 07:30.08 | shellshark | what version of asterisk? |
| 07:30.13 | Aurs | 1.2.13 |
| 07:30.16 | shellshark | odd |
| 07:30.31 | Aurs | call through a cisco sip gw |
| 07:30.41 | shellshark | ATA186? |
| 07:30.58 | dlynes_laptop | ata186 is an ata, not a gateway |
| 07:31.02 | Aurs | not sure, but I think it is something around 5000 |
| 07:31.08 | Aurs | 5400 or something |
| 07:31.33 | Aurs | oh, and no, it is not an ATA |
| 07:31.39 | shellshark | dlynes_laptop: ah, gateway would be something that translates SIP to SCCP in this case? |
| 07:31.53 | Aurs | it would be like using another asterisk |
| 07:32.08 | dlynes_laptop | shellshark: no...gateway is usually something that bridges two sip networks, or bridges fxo to fxs |
| 07:32.11 | Aurs | that cisco device has interfaces against pstn |
| 07:32.22 | shellshark | ah |
| 07:32.41 | dlynes_laptop | shellshark: cisco would never bridge sccp to sip...then they couldn't sell you sip licenses |
| 07:32.54 | shellshark | dlynes_laptop: they SELL sip? |
| 07:33.07 | dlynes_laptop | it's all about getting as many pounds of flesh out of their customer as possible :) |
| 07:33.16 | shellshark | well sure ;) |
| 07:33.22 | JT | lol, you think cisco give away firmware for free? ;) |
| 07:33.34 | shellshark | i thought you could download a SIP firmware for the devices that supported it at no charge |
| 07:33.41 | dlynes_laptop | shellshark: yeah...when you buy a cisco phone it comes with sccp software so you can only connect to their switch |
| 07:33.45 | *** join/#asterisk Mavvie (n=edwin@ppp21-59.lns2.syd7.internode.on.net) |
| 07:33.48 | shellshark | as long as you had a CCO account |
| 07:34.04 | shellshark | dlynes_laptop: chan_sccp is no good? |
| 07:34.09 | shellshark | or is it chan_skinny? |
| 07:34.16 | Aurs | cisco sells their mother if the price is right ;) |
| 07:34.18 | dlynes_laptop | shellshark: chan_sccp is crap...it hasn't been updated in quite some time |
| 07:34.25 | dlynes_laptop | shellshark: chan_skinny is what you should be using |
| 07:35.52 | dlynes_laptop | damnit |
| 07:35.59 | dlynes_laptop | seaweed crackers are all gone :((( |
| 07:36.11 | shellshark | chan_skinny is decent then? |
| 07:36.15 | dlynes_laptop | sure |
| 07:36.28 | Aurs | the calls that haven't been hung up has state "down" in show channels |
| 07:36.28 | dlynes_laptop | qwell/qwell[] is the author of it, too |
| 07:36.30 | shellshark | then why would people put SIP firmware on the phones anyway? |
| 07:36.54 | dlynes_laptop | shellshark: because it's more compatible with everything else and the cisco desktop manager is apparently buggy as all hell |
| 07:37.22 | shellshark | ah, i see |
| 07:37.33 | dlynes_laptop | and people would normally use that, not asterisk |
| 07:37.52 | shellshark | err wouldnt they normally use call manager or call manager express? |
| 07:37.59 | dlynes_laptop | erm clal manager...that's what it was called |
| 07:38.11 | dlynes_laptop | I don't use cisco, so I tend not to remember the names of their crap |
| 07:38.19 | shellshark | you still need call manager if you're using asterisk? |
| 07:38.34 | dlynes_laptop | I just know the ata186 because i've got two fo them collecting dust in a closet |
| 07:39.02 | shellshark | give me one :) |
| 07:39.11 | dlynes_laptop | what for? |
| 07:39.16 | shellshark | to play with |
| 07:39.20 | dlynes_laptop | the sipuras and linksys boxes are better quality |
| 07:39.28 | shellshark | i'm sure |
| 07:39.34 | shellshark | i use PAP2's all the time |
| 07:39.34 | dlynes_laptop | i'm serious |
| 07:39.41 | dlynes_laptop | the ata-186 was first generation |
| 07:39.46 | dlynes_laptop | the sipura 2000 was second generation |
| 07:39.47 | shellshark | but i'd like to play with an ATA186 or ATA188 |
| 07:39.51 | *** join/#asterisk [hC] (n=hardcore@S010600404521309c.vc.shawcable.net) |
| 07:39.51 | dlynes_laptop | and the pap2 is third generation |
| 07:39.53 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 07:40.26 | dlynes_laptop | besides |
| 07:40.35 | dlynes_laptop | i'm almost positive my boss wouldn't let it go for free |
| 07:40.41 | *** join/#asterisk XxxXX (i=Un1x@CPE000c419d026c-CM001225402bae.cpe.net.cable.rogers.com) |
| 07:40.46 | dlynes_laptop | not when he paid as much as he did for them |
| 07:41.00 | shellshark | ask him how much he'd want for one |
| 07:43.13 | dlynes_laptop | knowing him, probably about $200 USD :) |
| 07:45.28 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 07:45.28 | shellshark | chrap ;) |
| 07:45.34 | shellshark | -h |
| 07:45.41 | aadilismail | <PROTECTED> |
| 07:45.49 | aadilismail | what does it mean? |
| 07:46.57 | dlynes_laptop | aadilismail: it means you've got a codec incompatibility issue |
| 07:47.42 | dlynes_laptop | aadilismail: type 'show codecs' at the CLI to find out what type values the various codecs have |
| 07:48.29 | aadilismail | ok |
| 07:56.50 | *** join/#asterisk TonyM_ (n=TonyM@softins.claranet.co.uk) |
| 07:57.08 | *** part/#asterisk TonyM_ (n=TonyM@softins.claranet.co.uk) |
| 07:59.27 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:01.01 | *** join/#asterisk apardo_ (n=apardo@87.217.144.137) |
| 08:01.58 | Aurs | dlynes_laptop: soft hangup seems to have done the trick. thnx again |
| 08:02.38 | *** join/#asterisk af_ (n=af@ip-131-105.sn2.eutelia.it) |
| 08:11.22 | *** join/#asterisk frogzoo (n=frogzoo@202.155.165.25) |
| 08:20.10 | *** join/#asterisk lorinc (n=ang@caracas-2993.adsl.interware.hu) |
| 08:22.29 | *** join/#asterisk Sasch (n=Sasch@host102-30-static.107-82-b.business.telecomitalia.it) |
| 08:23.25 | *** join/#asterisk tekbasse (n=tekbasse@c-24-22-122-207.hsd1.or.comcast.net) |
| 08:26.33 | *** join/#asterisk petit-toon (n=petit-to@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 08:27.05 | *** join/#asterisk DerPraktikant (n=Tgu@p54B4D232.dip.t-dialin.net) |
| 08:27.06 | *** join/#asterisk zoa (n=d@pirus.securax.be) |
| 08:27.17 | DerPraktikant | hi m8´s |
| 08:28.30 | *** join/#asterisk [hC] (n=hardcore@S010600404521309c.vc.shawcable.net) |
| 08:33.06 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 08:36.48 | DerPraktikant | in which dir can i find the asterisk.conf |
| 08:42.19 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 08:48.33 | *** join/#asterisk apardo_ (n=apardo@87.217.146.127) |
| 08:54.39 | *** join/#asterisk hieunm_vips (n=hieunm_v@210.245.57.162) |
| 08:55.09 | *** join/#asterisk Tili (n=tili@202.133.67.238) |
| 08:56.58 | *** join/#asterisk grizzy (i=Generic@ppp-71-128-94-143.dsl.pltn13.pacbell.net) |
| 08:59.02 | *** join/#asterisk sergee (i=opera@195.94.224.197) |
| 09:01.19 | *** join/#asterisk jm|work (n=jamiem@dilbert.jamiem.com) |
| 09:07.41 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 09:07.41 | *** join/#asterisk don_ju (i=daemon@port-83-236-206-58.static.qsc.de) |
| 09:07.41 | *** join/#asterisk ftexcom (n=info@14.Red-80-26-4.staticIP.rima-tde.net) |
| 09:07.41 | hieunm_vips | Hi all |
| 09:07.42 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 09:08.10 | hieunm_vips | Can I create a application like as a wake-up clock |
| 09:09.02 | hieunm_vips | For ex: every morning, at 7 o'clock my phone will receive a call from Asterisk ! |
| 09:09.05 | *** join/#asterisk vatikan[16f] (n=miSafir@88.240.224.98) |
| 09:09.47 | *** join/#asterisk daysmen3 (n=primus@host81-158-207-70.range81-158.btcentralplus.com) |
| 09:11.18 | hieunm_vips | Or when I am home away, someone push the bell, I will receive the notify call |
| 09:11.30 | pif | tzafrir: hello, you package bristuff for debian? |
| 09:18.18 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
| 09:19.23 | *** join/#asterisk Ifaistos (n=stelios@ipa226.211.tellas.gr) |
| 09:19.40 | *** join/#asterisk dusan2 (n=dusan@209-223-47-160-static.oplink.net) |
| 09:21.05 | *** join/#asterisk ruskie (n=ruskie@sourcemage/mage/ruskie) |
| 09:21.38 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 09:27.24 | *** join/#asterisk apardo_ (n=apardo@87.217.144.167) |
| 09:27.50 | *** join/#asterisk Shaun2222 (n=ndci@ip68-5-62-187.oc.oc.cox.net) |
| 09:28.11 | Shaun2222 | i have a polycom 601 phone, is there a way to add a park button on the screen? |
| 09:29.15 | dlynes_laptop | Shaun2222: soft keys |
| 09:29.35 | Shaun2222 | ya actually i'm a idiot and just found it after pressing more.. |
| 09:29.46 | Shaun2222 | only problem is i pressed it and then pressed 1 and it hung up |
| 09:30.23 | Shaun2222 | also i dont see anythhing to pickup a parked call either... i so see pickup which i suppose may be for tha |
| 09:31.39 | dlynes_laptop | Shaun2222: parked calls is an asterisk feature, not a polycom feature |
| 09:31.50 | dlynes_laptop | Shaun2222: read up on parking lots on the wiki |
| 09:31.51 | *** join/#asterisk bmg505 (n=leon@c1-135-2.rndf.isadsl.co.za) |
| 09:32.08 | Shaun2222 | dlynes_laptop: right, well the polycom looks to have a "softbutton" for poarking calls |
| 09:32.16 | Shaun2222 | but apparently asterisk doesnt understand it |
| 09:32.22 | Shaun2222 | i have parking in asterisk enabled |
| 09:32.30 | dlynes_laptop | Shaun2222: yes, but you have to tell it what to send for parking a call |
| 09:32.46 | dlynes_laptop | Shaun2222: the default is probably not appropriate |
| 09:33.03 | dlynes_laptop | Shaun2222: or you might have to debug your dialplan to find out what it does send, and then modify your dialplan appropriately |
| 09:34.39 | *** join/#asterisk daysmen3 (n=primus@host217-44-109-89.range217-44.btcentralplus.com) |
| 09:34.51 | Shaun2222 | well asterisk isnt showing anything, just ending a call but i'm just watching with a -vvvvvvvvvv |
| 09:34.58 | tzafrir | pif, Debian packages bristuff, as well. I'm part of the team |
| 09:35.00 | Shaun2222 | i suppose i probably would have to do sip debug |
| 09:35.48 | dlynes_laptop | Shaun2222: or you could wait until someone comes on that actually knows something about polycom |
| 09:36.04 | dlynes_laptop | only things I know about polycom is what i've read on here from other people using them |
| 09:36.10 | Shaun2222 | well got a debug output |
| 09:36.22 | *** join/#asterisk Mw3 (i=mw3@national.t-error.hu) |
| 09:36.44 | Shaun2222 | Transfer to callpark in default |
| 09:37.00 | Shaun2222 | Transfer from 301 in default |
| 09:37.09 | Shaun2222 | SIP/2.0 603 Declined |
| 09:37.38 | *** join/#asterisk daysmen3 (n=primus@host217-44-109-89.range217-44.btcentralplus.com) |
| 09:38.20 | dlynes_laptop | Shaun2222: so you don't have callparking handled in asterisk, then |
| 09:38.24 | dlynes_laptop | nothing to do with the polycom |
| 09:38.32 | Shaun2222 | ya i do |
| 09:38.43 | Shaun2222 | but my context is called parkedcalls from the looks in features.conf |
| 09:38.49 | Shaun2222 | i think i may need to change it to callpark |
| 09:38.54 | *** join/#asterisk key2 (n=key2@251.9.39-62.rev.gaoland.net) |
| 09:39.13 | pif | tzafrir: I just sent a message to -users regarding a bristuff error "received SETUP message for call that is not a new call" |
| 09:39.32 | pif | have you seen that before? |
| 09:39.39 | dlynes_laptop | Shaun2222: check your features.conf file to see what the context is called for parked calls |
| 09:40.08 | dlynes_laptop | Shaun2222: then in extensions, make sure you include that context into whatever the outgoing context is for your polycom |
| 09:41.24 | Shaun2222 | hmm, that didnt do it.. what sucks, is when i hit park it wants me to put in somthing, i'm assuming a parking number and when i do, the other end heres me pressing keys |
| 09:41.29 | Shaun2222 | still failed though. |
| 09:41.53 | tzafrir | pif, hmm... no. It is indeed a message from libpri.patch |
| 09:42.28 | tzafrir | Would somebody please set up a bristuff mailing list? |
| 09:42.41 | dlynes_laptop | Shaun2222: you transfer to park, and then release the transfer |
| 09:42.48 | dlynes_laptop | Shaun2222: then you do a pickup from the other phone |
| 09:43.32 | Shaun2222 | ya i understand how it works... |
| 09:43.54 | Shaun2222 | i can park a call fine if i hit transfer, type in 700 wait for the system to tell me the parking lot number and hang up. |
| 09:44.02 | Shaun2222 | i was just hoping to get the softkey to work. |
| 09:44.37 | *** join/#asterisk florz_ (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 09:44.56 | dlynes_laptop | Shaun2222: it's probably just the configuration for that softkey then |
| 09:45.19 | pif | tzafrir: is kape still responsive to bug reports? |
| 09:46.55 | tzafrir | pif, that depends. It's a bit random, I guess |
| 09:47.36 | Shaun2222 | you'd think i could find documentation on configuring softkeys |
| 09:48.55 | dlynes_laptop | maybe in the polycom manual? |
| 09:48.56 | *** join/#asterisk xnon (i=xnon@200.82.222.118) |
| 09:50.50 | Shaun2222 | polycom manual isnt all that great... |
| 09:50.55 | Shaun2222 | they dont really give you much info. |
| 09:51.14 | Shaun2222 | they like to tell you about it but give you no example or syntax. |
| 09:51.43 | Shaun2222 | i think they expect you to use there default sip.cfg which has everything and enable/disable things... |
| 09:51.54 | Shaun2222 | only probelm is it doesnt show you how to change softkeys |
| 09:51.56 | pif | softkeys aren't configurable |
| 09:52.04 | pif | as far as I tried |
| 09:52.17 | Shaun2222 | pif: i guess that would make sense.. |
| 09:52.39 | Shaun2222 | pif, you dont happen to have SIP ver 2.0.2 or 2.0.3 do you? |
| 09:53.14 | *** join/#asterisk Tili (n=tili@202.133.67.133) |
| 09:53.24 | pif | http://www.freedomphones.net/polycom/files/ |
| 09:53.32 | Shaun2222 | they only have 2.0.1 |
| 09:53.52 | Shaun2222 | which i'm using.. |
| 09:54.25 | Aurs | isn't 2.0.1 the latest? |
| 09:54.38 | Shaun2222 | not according to polycom's site |
| 09:54.40 | Shaun2222 | they show 2.0.3 |
| 09:56.06 | sergee | Anybody willing to test new behaviour of attended transfer? |
| 09:56.19 | sergee | http://bugs.digium.com/view.php?id=8413 |
| 09:57.01 | *** join/#asterisk nettie (n=nettie@85-18-54-38.ip.fastwebnet.it) |
| 09:57.32 | Shaun2222 | thanks! |
| 09:58.44 | *** join/#asterisk daysmen3 (n=primus@host86-139-116-243.range86-139.btcentralplus.com) |
| 10:00.45 | *** join/#asterisk anthonyc (n=arcraig@adsl-75-10-114-228.dsl.frs2ca.sbcglobal.net) |
| 10:01.47 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 10:02.28 | *** join/#asterisk The_Ritz (n=rituraj@mail.pune.nevisnetworks.com) |
| 10:02.39 | nettie | I guys, since a couple of days I'm having big troubles with local phones registrations, call drops, and so on. I used to works properly before. I'm using Asterisk 1.2.12.1 with Polycom phones. The console outputs sipsock_read errors. The looks like network related problems. http://pastebin.ca/259297 Anyone have an idea of what coul dbe wrong please? The other funny behaviour is that when I start asterisk the sip module takes like 10+ seconds |
| 10:03.33 | Aurs | nettie: I also got those in > 1.2.7.1 |
| 10:03.38 | The_Ritz | can anyone guide me to wrtite a simple extensions.conf for skinny via two Cisco 7940 phones....i am sorry but i searched a lot and am pretty confused |
| 10:03.48 | Aurs | or >= 1.2.9 |
| 10:04.00 | Shaun2222 | anybody know with the poly phones if you can specify a bootimage name... |
| 10:04.49 | Aurs | Shaun2222: don't think you can specify name of bootrom.ld, but sip.ld can be specified in macaddr.cfg |
| 10:05.12 | nettie | Aurs that's incredible strange |
| 10:05.14 | Shaun2222 | ya i just set it for sip to be sip.ld.version |
| 10:05.18 | nettie | never had such problems before |
| 10:05.23 | Shaun2222 | was hoping to do the same with the bootrom |
| 10:05.29 | Aurs | nettie: I get them if i call someone, and hangup before I get answer |
| 10:05.47 | Aurs | Shaun2222: ok, don't know how (if possible) to do it with bootrom.. |
| 10:05.47 | nettie | Aurs nope mine are worse |
| 10:06.03 | Aurs | "BAD! BAD! BAD!" doesn't look so good, now does it? |
| 10:06.22 | nettie | Aurs I have random registry loss |
| 10:06.41 | Shaun2222 | now to test 2.0.3 |
| 10:06.43 | nettie | I know there's something wrong but I cant figure out what is it |
| 10:06.48 | *** part/#asterisk tekbasse (n=tekbasse@c-24-22-122-207.hsd1.or.comcast.net) |
| 10:06.54 | *** join/#asterisk jserve (n=mail@vpngw.de.callforcenetwork.eu) |
| 10:07.17 | jserve | Good morning |
| 10:07.26 | Aurs | nettie: I've had that too on polycoms in versions above 1.2.7.. tried to upgrade to 2.0.1 on polycom phones now, and haven't dropped reg yet |
| 10:07.38 | nettie | ahhhh |
| 10:07.39 | nettie | true |
| 10:07.47 | Aurs | (in asterisk versions above 1.2.7) |
| 10:07.59 | nettie | uhmm |
| 10:08.00 | nettie | well |
| 10:08.01 | Aurs | there is a nat keepalive in the 2.0 sip for polycom that me likes |
| 10:08.23 | nettie | maybe the polycom phones firmware then? |
| 10:08.36 | nettie | it's strange anyway |
| 10:08.46 | Aurs | yes it is.. I had no problems on 1.2.7.1 |
| 10:08.56 | Aurs | so something has changed |
| 10:09.12 | Shaun2222 | i still have to forward ports to get nat to work on mine... |
| 10:09.17 | nettie | I have 2 voip carrier |
| 10:09.17 | Shaun2222 | i can dial out, but nothing incomming |
| 10:09.25 | Shaun2222 | havnt tested 2.0 yet but i imagine it's the same. |
| 10:09.29 | nettie | if I remove on from sip.conf |
| 10:09.44 | nettie | on=one |
| 10:09.50 | Aurs | Shaun2222: in my experience, that depends on the nat router... |
| 10:09.50 | nettie | sip registry works |
| 10:10.05 | nettie | when I add it, everything doesnt register anymore |
| 10:10.07 | nettie | seems crazy |
| 10:10.28 | nettie | I add a carrier, local phones deregisters eheh |
| 10:10.35 | Aurs | what does "on=one" in sip.conf do? |
| 10:10.42 | nettie | no |
| 10:10.44 | nettie | was a type |
| 10:10.48 | nettie | on I meant one |
| 10:11.06 | Aurs | "one=no" ? |
| 10:11.21 | Shaun2222 | Aurs: the cisco's dont have a problem at all |
| 10:11.36 | Shaun2222 | i think the cisco's have stun in them? |
| 10:12.07 | Aurs | linksys sipura has nat keepalive. and they're working great |
| 10:12.09 | nettie | Aurs no |
| 10:12.16 | nettie | I wrote on instead of one |
| 10:12.17 | nettie | eheh |
| 10:12.19 | nettie | was a typo |
| 10:12.37 | Aurs | aaah. i get it |
| 10:12.37 | Aurs | hehe |
| 10:12.44 | Aurs | "if I remove one..." |
| 10:12.57 | Aurs | *inserting coffee* |
| 10:13.15 | nettie | eheh |
| 10:13.26 | nettie | well I'm downlaod polycom firmware 2.01 |
| 10:14.07 | Aurs | I get those "BAD! BAD! BAD!" messages, no matter what phones I am using |
| 10:14.17 | Aurs | (if I hangup before someone on the other end answers) |
| 10:14.30 | *** join/#asterisk knarfly (n=maxfiles@c-65-34-177-3.hsd1.fl.comcast.net) |
| 10:15.11 | Aurs | but only 1 time pr call |
| 10:16.01 | nettie | nope, I'm definitely having registration problems when I enable multiple carriers |
| 10:16.26 | *** part/#asterisk marloshome (n=markpetr@76.184.193.52) |
| 10:18.17 | Aurs | nettie: interresting.. |
| 10:19.22 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 10:22.23 | SheriF_SpacE | how can i detecet the busy / hangup on zap "FXO module " which connected to a line coming from Cisco VG :-s? |
| 10:22.44 | *** join/#asterisk henrique (n=henrique@201-1-131-176.dsl.telesp.net.br) |
| 10:23.51 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 10:26.27 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 10:31.46 | The_Ritz | I get |
| 10:31.55 | The_Ritz | "trying to send ' ' " |
| 10:32.05 | The_Ritz | displaying ' ' |
| 10:32.10 | The_Ritz | disconnected from asterisk |
| 10:32.25 | The_Ritz | i am using cisco 7940 + skinny |
| 10:34.39 | *** join/#asterisk ontae (n=ontae@clnet-p03-090.ikbnet.co.at) |
| 10:35.20 | ontae | hi folks, nay |
| 10:44.19 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 10:45.41 | *** join/#asterisk ruskie (n=ruskie@sourcemage/mage/ruskie) |
| 10:47.38 | *** join/#asterisk ontae (n=ontae@clnet-p03-090.ikbnet.co.at) |
| 10:52.30 | Shaun2222 | anybody know how to use the speed dial on the polycoms |
| 10:56.30 | SheriF_SpacE | zaptel ignrouing my FXS modules :-s |
| 10:57.09 | *** join/#asterisk Tili (n=tili@202.133.67.26) |
| 11:02.18 | *** join/#asterisk Fraeggl (n=Fraeggl@rkom.r-kom.de) |
| 11:03.10 | dlynes_laptop | SheriF_SpacE: do you have your power plugged into your digium or sangoma card? |
| 11:06.30 | Aurs | Shaun2222: up arrow |
| 11:07.20 | Aurs | or line keys |
| 11:07.36 | *** join/#asterisk shadebob (n=chatzill@84.16.31.10) |
| 11:08.33 | shadebob | hi, I try to use the sendtext cmd in the CLI but I have a message "not in a call". Someone can help me to use send text? |
| 11:11.22 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 11:11.59 | SheriF_SpacE | http://pastebin.ca/259333 |
| 11:12.02 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:12.05 | SheriF_SpacE | dlynes_laptop: yes check out http://pastebin.ca/259333 |
| 11:12.43 | SheriF_SpacE | if i removed the FXS zaptel works |
| 11:13.52 | The_Ritz | I am using RH9 for skinny + cisco 7940 |
| 11:14.00 | *** part/#asterisk hieunm_vips (n=hieunm_v@210.245.57.162) |
| 11:14.21 | dlynes_laptop | SheriF_SpacE: what's your zaptel.conf look like? |
| 11:14.34 | dlynes_laptop | SheriF_SpacE: and i'm guessing you're using digium hardware, right? |
| 11:14.39 | SheriF_SpacE | yes |
| 11:14.42 | SheriF_SpacE | TDM400P |
| 11:15.31 | dlynes_laptop | SheriF_SpacE: pastebin your zaptel.conf file |
| 11:15.35 | SheriF_SpacE | http://pastebin.ca/259336 |
| 11:15.41 | The_Ritz | when I call from one phone to another, it rings .....the moment i pickup the call the kernel crashes |
| 11:15.47 | The_Ritz | it this known? |
| 11:15.55 | *** join/#asterisk apardo (n=apardo@62.15.239.65) |
| 11:16.23 | dlynes_laptop | SheriF_SpacE: ummmmm |
| 11:16.49 | dlynes_laptop | SheriF_SpacE: You mind putting some commenting in there so that someone that didn't write your config files can figure out which is which? |
| 11:17.13 | SheriF_SpacE | dlynes_laptop: and loadzone = us |
| 11:17.13 | dlynes_laptop | SheriF_SpacE: also, can you pastebin your dmesg? |
| 11:17.14 | SheriF_SpacE | defaultzone=us |
| 11:17.32 | SheriF_SpacE | okay |
| 11:17.35 | dlynes_laptop | SheriF_SpacE: copy/paste it to pastebin; don't type it in |
| 11:17.52 | SheriF_SpacE | ProSLIC on module 0 failed to powerup within 501 ms (0 mV only) |
| 11:17.52 | SheriF_SpacE | -- DID YOU REMEMBER TO PLUG IN THE HD POWER CABLE TO THE TDM400P?? |
| 11:17.56 | SheriF_SpacE | oh god ! |
| 11:18.00 | SheriF_SpacE | wait may be really i did forget ! |
| 11:18.05 | SheriF_SpacE | brb |
| 11:26.59 | *** join/#asterisk CleanerX (n=nix@p54A391F2.dip0.t-ipconnect.de) |
| 11:27.12 | *** join/#asterisk ruskie (n=ruskie@sourcemage/mage/ruskie) |
| 11:27.16 | SheriF_SpacE | dlynes_laptop: i was sure i connected it but it was another machine :-) not this one :-D |
| 11:27.41 | *** join/#asterisk jay (n=jay@lindalane.com) |
| 11:27.44 | dlynes_laptop | good job :) |
| 11:33.19 | *** join/#asterisk tinpot (n=nknight@217.145.120.198) |
| 11:33.34 | tinpot | morning all |
| 11:34.47 | *** join/#asterisk coppice (n=chatzill@40.168.17.210.dyn.pacific.net.hk) |
| 11:35.08 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 11:35.29 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 11:35.44 | tinpot | question for you all present, how important do you think the "one-device" i.e. mobile phone come corporate phone (camped on the corporate VOIP network) will be? |
| 11:47.11 | *** join/#asterisk marexz (n=marexz@195.20.127.222) |
| 11:47.38 | HarryR | tinpot, give it another 5 years |
| 11:48.07 | HarryR | have to wait for the bigwigs to realize you can get naughty content on them before they'll become widespread |
| 11:49.22 | *** join/#asterisk christofer (n=christof@nat130.td1mobile.de) |
| 11:49.49 | christofer | . |
| 11:50.10 | *** join/#asterisk Skarmeth (n=Skarmeth@201009008073.user.veloxzone.com.br) |
| 11:50.14 | christofer | has someone still installed asterisk on an 1und1.de rootserver? |
| 11:50.33 | *** join/#asterisk lorinc (n=ang@caracas-2993.adsl.interware.hu) |
| 11:52.34 | *** join/#asterisk yel (n=yassine@xdsl-87-78-23-249.netcologne.de) |
| 11:55.46 | *** join/#asterisk merbanan (n=Anders@136.240.13.217.in-addr.dgcsystems.net) |
| 12:04.52 | dlynes_laptop | Is there a way that I could play back a sound file to a caller, and while it's doing that, make another call in the background which I then playback a file to after it answers? |
| 12:05.15 | dlynes_laptop | Or is asterisk even capable of something like that? |
| 12:05.31 | merbanan | dlynes_laptop: look up call files |
| 12:05.47 | dlynes_laptop | Yeah...using one, but it's not working for what I need |
| 12:06.00 | dlynes_laptop | Trying to use one to send out faxes |
| 12:07.29 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 12:07.37 | *** join/#asterisk vieri (n=rentorbu@201.Red-213-96-91.staticIP.rima-tde.net) |
| 12:08.20 | Winkie | hey guys, what exactly is overlap dialling? |
| 12:08.28 | Winkie | because i'm not sure if i need it in any way |
| 12:08.30 | dlynes_laptop | merbanan: do you perhaps have a pointer to some good documentation on call files? |
| 12:10.14 | merbanan | dlynes_laptop: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out <- best I could find |
| 12:10.28 | vieri | I was wondering if someone could take a look at http://forums.digium.com/viewtopic.php?p=37514#37514 I'm having trouble with Flash() only when using queues. |
| 12:10.31 | dlynes_laptop | merbanan: much appreciated...thanks |
| 12:12.10 | SheriF_SpacE | hmmmm i hate zaptel :( |
| 12:12.25 | SheriF_SpacE | dlynes_laptop: any idea ? http://pastebin.ca/259371 |
| 12:13.03 | dlynes_laptop | SheriF_SpacE: looks fine to me, except for the configuration parameters you're trying to give your tdm400p that aren't valid |
| 12:14.00 | SheriF_SpacE | there is no more signalling ? |
| 12:14.25 | dlynes_laptop | SheriF_SpacE: one sec, and i'll paste a sample zaptel.conf/zapata.conf file for you |
| 12:14.38 | SheriF_SpacE | okay |
| 12:19.18 | dlynes_laptop | SheriF_SpacE: http://pastebin.ca/259375 |
| 12:19.37 | dlynes_laptop | SheriF_SpacE: Those config files are for four fxo ports |
| 12:19.57 | dlynes_laptop | SheriF_SpacE: The third and fourth port are not being used, so they're commented out in the zapata.conf file |
| 12:20.25 | *** join/#asterisk pavel_s (i=pi@195.22.146.120) |
| 12:20.35 | dlynes_laptop | Pavel Bure! |
| 12:22.39 | SheriF_SpacE | dlynes_laptop: nothing wrong with mine then :-s |
| 12:22.54 | SheriF_SpacE | why it don't ignore singalling !? |
| 12:23.25 | dlynes_laptop | Repost your latest zaptel.conf and zapata.conf files |
| 12:23.32 | dlynes_laptop | and use copy/paste to put it into pastebin |
| 12:23.40 | dlynes_laptop | don't type it in again, please |
| 12:24.13 | SheriF_SpacE | okay i'm sorry :-) |
| 12:32.42 | *** join/#asterisk pavel_s (i=pi@195.22.146.120) |
| 12:33.00 | *** join/#asterisk nafigator (n=IM@84.22.192.250) |
| 12:35.55 | *** join/#asterisk zeppelin_ (n=zeppelin@201.66.140.68) |
| 12:36.03 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 12:36.42 | puzzled | hi all |
| 12:39.01 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 12:41.45 | *** join/#asterisk tr2x (n=alvar@80-218-150-90.dclient.hispeed.ch) |
| 12:42.47 | *** join/#asterisk aztekrw (n=aztekrw@dsl-244-195-89.telkomadsl.co.za) |
| 12:45.59 | aztekrw | I have a question about using Origionate from the manager, and transfers |
| 12:47.37 | aztekrw | I origionate a call from from a sip phone to a conference room that already has a call that came in from my PRI line. |
| 12:48.09 | aztekrw | but the SIP phone that answers the call, and is put into the conference with the call from the PRI cannot transfer that call on to somebody else. |
| 12:48.21 | aztekrw | anybody have a solution for me? |
| 12:48.44 | *** join/#asterisk Flosoft (n=admin@d51A474D9.access.telenet.be) |
| 12:54.32 | *** join/#asterisk alerios (n=alerios@201.244.173.92) |
| 12:54.33 | *** join/#asterisk FreezeS (n=Gladius@82.208.156.94) |
| 12:56.42 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 12:57.29 | FreezeS | hey guys |
| 12:57.48 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 12:57.53 | FreezeS | I want to do something like a cluster, with users being able to register to multiple IPs |
| 12:58.10 | FreezeS | how can I make a queue find a user if it's registered to another computer ? |
| 13:01.01 | dlynes_laptop | wow...might have finally gotten a fax to complete transmission :) |
| 13:01.40 | *** join/#asterisk w9sh (n=w9sh@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 13:03.40 | *** join/#asterisk Mother_ (n=mother@93.Red-80-32-127.staticIP.rima-tde.net) |
| 13:05.00 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-166-226.buff.east.verizon.net) |
| 13:05.33 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 13:05.54 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 13:06.05 | *** join/#asterisk folsson (n=folsson@h24n2fls34o985.telia.com) |
| 13:08.34 | *** join/#asterisk iCEBrkr (i=icebrkr@69.9.167.70) |
| 13:09.33 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 13:10.27 | dlynes_laptop | bleh....but only 1 page out of 5 :((( |
| 13:10.38 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 13:11.07 | *** join/#asterisk steveaj (n=sj@62.55.147.53) |
| 13:11.50 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 13:12.02 | *** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net) |
| 13:12.14 | shmaltz | does the aastra phone support auto answer? |
| 13:12.19 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:12.32 | Ifaistos | FreezeS : Go to http://www.astricon.net/files/usa06/Friday-General_Conference/ and download the two papers from JR Richardson |
| 13:12.51 | dlynes_laptop | shmaltz: yes |
| 13:12.55 | Ifaistos | FreezeS : That's one way to do it |
| 13:13.01 | shmaltz | ty dlynes_laptop |
| 13:13.13 | hi365 | is there anyway to have asterisk print out a fax via a network printer? |
| 13:13.16 | aadilismail | hi |
| 13:13.16 | dlynes_laptop | shmaltz: the 9133i's do, and hte 480i's do |
| 13:13.27 | dlynes_laptop | shmaltz: i don't know it the 9118's do or not |
| 13:13.40 | *** join/#asterisk dioedu (n=dioedu@201.7.117.114) |
| 13:13.40 | shmaltz | I'm talking about the 480is |
| 13:13.42 | dlynes_laptop | hi365: sure, why not? |
| 13:13.43 | aadilismail | can i check complete call history .... |
| 13:13.53 | hi365 | dlynes_laptop: how so? |
| 13:14.03 | aadilismail | what is the call history command ? |
| 13:14.09 | dlynes_laptop | hi365: if you can convert the fax to a pdf and email it out, i'm sure you can print it |
| 13:14.13 | *** join/#asterisk Tili (n=tili@202.133.67.142) |
| 13:14.21 | dlynes_laptop | hi365: convert the fax to a postscript document and get ghostscript to print it |
| 13:15.00 | dlynes_laptop | hi365: how is all going to depend on what printer you're using, and what drivers you've got installed for it...i.e. lp, lpd, lpr, cups, samba, ... |
| 13:15.12 | *** join/#asterisk dasenjo (n=dasenjo@63.245.86.186) |
| 13:15.43 | hi365 | dlynes_laptop: u lost me a bit. ive got a windows network shared printer so i guess thats going to b samba |
| 13:16.02 | dioedu | hello, does someone know something about deadlock's in chan_local ? I have A LOT OF deadlock's in this channels that i'm using in my queues. Sometimes i have deadlock's in chan_sip. |
| 13:16.03 | dlynes_laptop | hi365: but basically, if you know what the command is to print to your machine, you just have to convert the tiff to a ps file, and then issue your print command |
| 13:16.11 | dlynes_laptop | hi365: correct |
| 13:16.21 | FreezeS | thanks Ifaistos |
| 13:16.28 | dlynes_laptop | hi365: if you want to know how to print to it, i would check on #samba |
| 13:16.29 | hi365 | dlynes_laptop what file do i set that from? |
| 13:16.41 | dlynes_laptop | hi365: but probably more appropriate would be the samba mailing lists |
| 13:16.51 | dlynes_laptop | hi365: it's not something someone can answer in a couple of seconds |
| 13:17.10 | hi365 | for the samaba part. but where do i set the * to print to a tiff? |
| 13:17.35 | dlynes_laptop | hi365: app_rxfax --> http://www.soft-switch.org/ |
| 13:17.47 | dlynes_laptop | hi365: make sure you're not running a version of asterisk newer than 1.2.9.1, too |
| 13:17.57 | dlynes_laptop | hi365: anything newer than that may or may not work |
| 13:18.27 | hi365 | brb |
| 13:19.39 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 13:19.53 | *** join/#asterisk zigman (i=zigman@irc.zigman.de) |
| 13:20.27 | *** join/#asterisk frogzoo (n=frogzoo@202.155.165.25) |
| 13:20.40 | aadilismail | can i check complete call history record ? wht is the command ??/? |
| 13:21.03 | dlynes_laptop | aadilismail: cat /var/log/asterisk/cdr-csv/Master.csv |
| 13:25.04 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 13:28.17 | *** join/#asterisk upod1 (n=chibondk@cpe-66-108-211-222.nyc.res.rr.com) |
| 13:32.04 | *** join/#asterisk af_ (n=af@ip-131-105.sn2.eutelia.it) |
| 13:33.01 | *** join/#asterisk merbanan (n=Anders@136.240.13.217.in-addr.dgcsystems.net) |
| 13:36.38 | aadilismail | ANSWERED DOCUMENTATION mean ? |
| 13:37.07 | *** join/#asterisk RoyK (n=roy@217-175-34.100710.adsl.tele2.no) |
| 13:37.16 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 13:39.01 | dlynes_laptop | check the documentation for "call disposition" and "ama flags" on voip-info.org |
| 13:39.06 | *** part/#asterisk dasenjo (n=dasenjo@63.245.86.186) |
| 13:39.16 | af_ | what choices I have to manage faxes with *? |
| 13:40.17 | dlynes_laptop | af_: do a search for 'fax' on www.voip-info.org |
| 13:40.24 | dlynes_laptop | af_: you'll find a bunch of stuff |
| 13:40.40 | dlynes_laptop | af_: most of it uses spandsp under the hood though...spandsp is available at www.soft-switch.org |
| 13:41.05 | *** part/#asterisk muehlbucks (n=muehlbuc@24-148-29-44.mct-bsr1.chi-mct.il.static.cable.rcn.com) |
| 13:41.20 | af_ | oh ok. any other thigs other then spandsp based? |
| 13:41.38 | dlynes_laptop | asterfax might not be spandsp based |
| 13:41.46 | dlynes_laptop | but i think it's probably commercial, too |
| 13:41.47 | af_ | oh ok, thanks so much |
| 13:42.20 | af_ | oh, another thing. any distribution * ready that does faxing in/out? |
| 13:44.53 | hi365 | dlynes_laptop: im back. thanks for the info |
| 13:45.11 | hi365 | just 1 question. is there a config file with fax settigns for *? |
| 13:45.12 | dlynes_laptop | np |
| 13:46.01 | coppice | they're all based on spandsp. there is no other soft-fax solution that fits into * |
| 13:49.48 | *** join/#asterisk glasco36 (n=dasver96@c-69-254-230-163.hsd1.ks.comcast.net) |
| 13:49.55 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 13:51.45 | dlynes_laptop | hi365: no |
| 13:52.55 | dlynes_laptop | hi365: if you're wanting something like that, I would suggest taking a look at hylafax...it's got a nice, easy way to set up cover pages, give your windows machines a fax printer, ... |
| 13:53.05 | *** join/#asterisk dasenjo (n=dasenjo@201.228.128.10) |
| 13:53.14 | dlynes_laptop | hi365: then you can run that through iaxmodem (part of spandsp) to fax out on asterisk |
| 13:53.17 | *** join/#asterisk exarverspuy (n=root@h8441179167.dsl.speedlinq.nl) |
| 13:53.58 | dlynes_laptop | oops..my mistake |
| 13:54.07 | dlynes_laptop | i guess it isn't part of spandsp...it just uses spandsp |
| 13:54.18 | dlynes_laptop | sf.net/projects/iaxmodem |
| 13:55.05 | coppice | iaxmodem is a wrapper around spandsp to connect it into hylafax |
| 13:55.51 | *** part/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 13:56.21 | exarverspuy | Hi All, I've a question about the g729 codec of Digium... |
| 13:56.36 | exarverspuy | I'm running asterisk on a Pentium D x86_64. |
| 13:56.55 | exarverspuy | But I don't know exactly which version to use from the digium ftp site for the g729 codec... |
| 13:57.07 | exarverspuy | anybody here who does know which one to use? |
| 13:58.14 | aadilismail | what is the meaning of "ANSWERED" "DOCUMENTATION"??? during checking CLI ? |
| 13:58.35 | dlynes_laptop | aadilismail: scroll back up the screen; I already answered your question |
| 13:59.12 | aadilismail | SORRY |
| 14:05.02 | *** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no) |
| 14:05.31 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 14:09.23 | exarverspuy | Are there any Digium g729 codec guru's around her? Or am I too early? |
| 14:09.41 | *** join/#asterisk ontae (n=ontae@clnet-p03-090.ikbnet.co.at) |
| 14:10.17 | ontae | Hi folks, may someone help me with a "401 Unauthorized" ? |
| 14:10.38 | dlynes_laptop | ontae: it means your username and/or password is incorrect |
| 14:14.03 | ontae | dlynes_laptop: username/password ist O.K.! I get from my "Grandstream BT110 1.0.8.33" the following syslog_entry: "401 Unauthorized ... WWW-Authenticate: Digest algorithm=MD5, realm="<DOMAIN>", nonce="099d4637" Content-Length: 0 |
| 14:15.03 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool1-11.mtco.com) |
| 14:15.31 | dlynes_laptop | ontae: exactly...username and/or password is not correct |
| 14:16.11 | dlynes_laptop | What's a BT-110? |
| 14:16.46 | dlynes_laptop | Is it an upgraded version of the BT-101 or BT-102? |
| 14:16.56 | *** join/#asterisk jaike (n=jaike@210.213.125.121) |
| 14:18.36 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 14:18.58 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:20.22 | *** join/#asterisk Laerte (n=io@217.221.36.10) |
| 14:20.44 | *** join/#asterisk M_at (n=643744f8@82.152.60.82) |
| 14:21.32 | *** join/#asterisk Dio_ (n=dima@arkadia.soborka.net) |
| 14:21.34 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 14:22.25 | *** join/#asterisk b11d (n=noway@234-200-29-134.hcc.mnscu.edu) |
| 14:22.35 | b11d | morning everyone |
| 14:22.44 | M_at | afternoon |
| 14:23.45 | b11d | heh |
| 14:23.51 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 14:23.59 | b11d | i need an eye opener.. |
| 14:24.24 | b11d | thanks :) |
| 14:24.34 | b11d | thats just what I was thinking abouyt |
| 14:26.11 | aadilismail | can i check the specific number CLI... or call history ? |
| 14:26.45 | dlynes_laptop | aadilismail: did you read my previous answer? |
| 14:27.21 | aadilismail | ok |
| 14:27.22 | aadilismail | sorry |
| 14:28.59 | Dio_ | hello, anybody can help me with disclaimer? |
| 14:29.03 | *** join/#asterisk qwertz_ (n=qwertz@pD9531926.dip0.t-ipconnect.de) |
| 14:30.35 | M_at | Your house may be at risk if you drive a JCB through it? |
| 14:30.48 | *** join/#asterisk Alex_112 (n=admin@fw.packetfront.com) |
| 14:31.08 | dlynes_laptop | Japanese bank card? |
| 14:31.14 | Laerte | anyone has problem with attendant transfer & cdr logging ? |
| 14:32.13 | Alex_112 | what is the variablename of the number that has been dialed |
| 14:32.17 | Alex_112 | ? |
| 14:32.40 | dlynes_laptop | ${EXTEN} |
| 14:33.07 | ontae | dlynes_laptop: I upgraded my BT-102 last weekend to Firmware 1.0.8.33 and it seems as it reports itself to syslog as "BT110", strange ! |
| 14:33.19 | *** join/#asterisk pingwin (n=pingwin@216.249.143.62) |
| 14:33.56 | dlynes_laptop | ontae: the 'username' on the grandstream equates to the name in '[...]' in the sip.conf file |
| 14:34.27 | dlynes_laptop | ontae: you don't need to specify username= in sip.conf, and you still need secret=... whatever the value was you picked for your password on the phone |
| 14:34.47 | qwertz_ | Hi, I've got problem to get music on hold working. mpg123 is installed the mp3 directory and its file is rwx. directory=/var/lib/asterisk/mohmp3/default is set in musiconhold.conf - but still nothing. Any hints where I could take a further look? |
| 14:35.08 | Alex_112 | dlynes_laptop, tnx |
| 14:35.45 | dlynes_laptop | anyways |
| 14:35.53 | dlynes_laptop | on that note, i'm hitting the hay |
| 14:36.01 | b11d | bye bye |
| 14:36.30 | b11d | qwertz_.. so how are you calling MoH from your extensions.conf? |
| 14:37.56 | ontae | dlynes_laptop: 'username' in phone = [username] in sip.conf, no username= in that section and secret= is set correctly, i verified my username/password several times, may the 401 can have onother cause? |
| 14:38.36 | b11d | ontae.. can you ping each phone? |
| 14:39.47 | ontae | b11d: if you mean ICMP Ping - no, because ist resides behind a firewall; but what is your intension about that question? |
| 14:40.03 | b11d | uhh.. in resolving your 401 |
| 14:40.10 | b11d | and the word is "intention" :) |
| 14:40.17 | *** join/#asterisk Juggie (n=Juggie@wiley-459-29776.roadrunner.nf.net) |
| 14:40.35 | xnon | anybody can tel me about any good softphone that isnt XLITE |
| 14:40.40 | b11d | there are none |
| 14:40.46 | pingwin | kphone? |
| 14:41.43 | *** part/#asterisk izmarkie (n=iz-dluv@64.186.61.162) |
| 14:41.56 | qwertz_ | b11d, have set "musicclass=default" in my sip.conf |
| 14:42.15 | b11d | hmmm |
| 14:42.23 | ontae | b11d: ya, meant intention - solving my 401 would be great |
| 14:42.31 | b11d | and no audio at all? |
| 14:42.55 | b11d | I doubt I can help you ontae.. never worked on NAT issues before, sorry. |
| 14:43.39 | ontae | b11d: Why do you think this is a NAT issue? |
| 14:44.01 | b11d | because you're talking about 401 errors and you use NAT.. i'd guess NAT is getting in your way. |
| 14:44.29 | b11d | NAT & SIP dont play well together.. AT ALL. |
| 14:44.35 | qwertz_ | b11d, MP3Player(/var/lib/asterisk/mohmp3/default/weekend.mp3 plays without problems ... |
| 14:44.44 | b11d | aparently you can do some things with STUN though to overcome them |
| 14:44.56 | b11d | qwertz_.. yeah but we're not working on that are we :) |
| 14:45.07 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 14:46.08 | ontae | yes, NAT is in my way between asterisk and one UA (Grandstream BT102) which gets a 401, but i have no probelm with another SPA942 (also NAT between) |
| 14:46.50 | M_at | Depends on the NAT device - some version of Sonicwall's software, for instance, do SIP transformations through their NAT |
| 14:47.54 | pingwin | hey guy's I've got a big problem. I'm using asterisk to operate as a pbx for a esf/b8zs PRI line. I do not use NAT. But about every 15 minutes the chan_zap.c throws RED alarms on all channels begining with error stating "PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1" |
| 14:47.59 | pingwin | can ANYONE help? |
| 14:48.16 | pingwin | i've already adjusted the IRQ's and changed the PCI slot the card sits in |
| 14:48.38 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:48.38 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:49.07 | pingwin | this was working fine for over a month. and now this. |
| 14:49.23 | coppice | I wannabe the first one on the block to do G.729.1 :-) |
| 14:49.48 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 14:49.50 | bkw__ | coppice, but you gotta watch out for those pesky interrupt issues. |
| 14:51.07 | *** join/#asterisk upod (n=chibondk@cpe-66-108-211-222.nyc.res.rr.com) |
| 14:56.56 | b11d | ping.. call your provider? |
| 14:57.13 | b11d | if it just stopped working, chances are its the telco end.. |
| 14:57.18 | *** part/#asterisk tinpot (n=nknight@217.145.120.198) |
| 14:57.22 | pingwin | well I have. |
| 14:57.24 | b11d | if its not the telco end, then it could be your card.. call your tech support line for that |
| 14:57.31 | b11d | for warranty info :) |
| 14:57.46 | pingwin | the t1 has been tested and everything seems fine, and our PRI provider says everything is fine and they receive no errors or flags, even when we crash |
| 14:57.50 | b11d | did you replace your t1 cable? |
| 14:58.12 | b11d | what kind of pri card? |
| 14:58.26 | pingwin | no, it's kind of a process for us to do that. it's on the other side of the building inside of a datacenter |
| 14:58.35 | pingwin | pri card is a digium T100P |
| 14:58.36 | b11d | well.. go do that.. |
| 14:58.39 | pingwin | single span |
| 14:58.42 | b11d | bad cables DO happen |
| 14:58.52 | b11d | rarely though |
| 14:58.57 | pingwin | could that be creating these HDLC errors? |
| 14:59.01 | b11d | yes it could |
| 14:59.16 | b11d | especially if they "just started" out of nowhere |
| 14:59.24 | b11d | or.. call Digium about replacing the card.. |
| 14:59.41 | b11d | this doesnt sound like an asterisk issue, especially since you changed nothing.. and this just stopped working all of a sudden. |
| 15:00.00 | pingwin | yeah I don't think it's an asterisk issue either |
| 15:00.10 | pingwin | just need help :) |
| 15:00.13 | b11d | call digigum and/or replace cable |
| 15:00.19 | pingwin | thanks b11d |
| 15:00.29 | b11d | no prob.. i just hope that helps in some way |
| 15:00.42 | b11d | I just dont know where else to look as your telco says their end is fine. |
| 15:01.00 | b11d | though, if they dont see the crash on the end, that is suspicious as well.. |
| 15:01.10 | b11d | they should see the link drop or something |
| 15:01.12 | b11d | you'd think |
| 15:01.14 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 15:04.28 | b11d | ping.. you read this? http://www.asteriskguru.com/tutorials/hdlc_bad_fcs.html |
| 15:08.35 | *** join/#asterisk tegioz (n=tegioz@62.87.55.130) |
| 15:10.26 | *** join/#asterisk ManxPower (n=manxpowe@71-8-11-73.dhcp.leds.al.charter.com) |
| 15:10.45 | *** join/#asterisk Blaet (n=rogier@82-204-26-196.dsl.bbeyond.nl) |
| 15:10.50 | Blaet | Hi all |
| 15:11.25 | Blaet | When compiling the lastest stable version of asterisk I get an error that it can't find lssl |
| 15:11.33 | Blaet | Can somebody tell me which library I'm missing? |
| 15:12.48 | SuPrSluG | ssl-devel |
| 15:12.55 | Ifaistos | Blaet : libssl-dev |
| 15:13.29 | *** join/#asterisk ruskie (n=ruskie@sourcemage/mage/ruskie) |
| 15:14.22 | Blaet | ok thanks |
| 15:17.41 | *** join/#asterisk saftsack (n=saftsack@pD9E04945.dip.t-dialin.net) |
| 15:19.48 | Dio_ | any bugmarshals around? |
| 15:19.50 | ontae | M_at: your right, SIP transformation can happen through NAT devices |
| 15:20.31 | M_at | But only if the NAT device specifically support it |
| 15:21.06 | ManxPower | M_at: that is incorrect. |
| 15:21.37 | ManxPower | As pretty much any SIP ITSP shows. |
| 15:22.13 | M_at | No - SIP transformation - not SIP passing through. |
| 15:22.21 | ManxPower | SIP support in NAT routers is so shoddy, inconsistent, and bug ridden that I normally disable that in the router and rely on Asterisk |
| 15:22.37 | ManxPower | M_at: define "SIP transformation" |
| 15:22.41 | M_at | If SIP isn't transformed then you need a STUN server or similar, if it is transformed by the NAT DEvice it negates the need for the STUN server |
| 15:23.12 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 15:23.22 | ManxPower | M_at: I have run SIP thru many NAT devices without any need for STUN or other hack. |
| 15:23.24 | M_at | I'm not claiming that SIP doesn't work through NAT and can only work through NAT with the right kit - just that SIP transformation at the NAT device can make things alot easier |
| 15:23.56 | M_at | My original comment was "Depends on the NAT device - some version of Sonicwall's software, for instance, do SIP transformations through their NAT" |
| 15:24.27 | ManxPower | M_at: and I'm saying that in my experience SIP NAT "transformation" support is so BAD in most devices it actually causes more problems than it solves. |
| 15:24.33 | *** join/#asterisk __AK__ (n=ak@28.228.210.62.te-dns.org) |
| 15:24.42 | pif | hello, has anyone seen that bristuff error "received SETUP message for call that is not a new call" ? |
| 15:25.12 | M_at | Then why tell me that I am incorrect? |
| 15:25.27 | M_at | <ontae> M_at: your right, SIP transformation can happen through NAT devices |
| 15:25.32 | M_at | <M_at> But only if the NAT device specifically support it |
| 15:25.45 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.232.28.Dial1.SanJose1.Level3.net) |
| 15:25.48 | ManxPower | M_at: To make SIP worth thru NAT when the client is behind NAT and Asterisk is not: put nat=yes in sip.conf. Done. |
| 15:25.51 | M_at | Is true - SIP will not be transformed through NAT if the NAT device does not offer that facility |
| 15:26.23 | pingwin | b11d thanks for that link, I'll give it a try |
| 15:26.35 | Dio_ | anybody can help me with disclaimer? |
| 15:26.45 | ManxPower | To make SIP work thru NAT when Asterisk is behind NAT: set localnet and externip in sip.conf, set rtp.conf, port forward the ports in rtp.conf and 5060. Done. |
| 15:27.01 | ManxPower | Dio_: there are TWO disclaimers that you can use. |
| 15:28.04 | ManxPower | Dio_: One of them just assigns all rights to Digium. The other one just grants an unrestricted license to your code, but you keep the copyright. |
| 15:28.06 | Dio_ | ManxPower: I know that, I already fill one of them and need to submit it to Digium. But I don't have FAX. Some guy on bug tracker told me I can email signed PDF. The question is where should I sent it? |
| 15:28.24 | ManxPower | Dio_: No idea. |
| 15:28.44 | M_at | And did I ever say that the only way SIP would work through NAT was with NAT device based translations? NO! I just said that some devices perform the translations. For your next trick will you teach your mother to suck eggs? |
| 15:29.13 | ManxPower | (09:22:37) M_at: If SIP isn't transformed then you need a STUN server or similar, if it is transformed by the NAT DEvice it negates the need for the STUN server |
| 15:29.28 | Dio_ | ManxPower: oh.. |
| 15:29.32 | ManxPower | That is what you said. 8-) |
| 15:29.44 | M_at | "or similar" would count for configuring Asterisk to cope |
| 15:29.45 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 15:30.16 | ManxPower | M_at: many, many, many people thing that NAT requires STUN, etc. They say it all the time. I'm just trying to make sure that people know that is not true. |
| 15:30.26 | M_at | The original comments were made in relation to a guy with one handset working through NAT and another not - my comment was meant to indicate that there was one possible reason for that. There are, of course, many |
| 15:30.29 | ManxPower | M_at: Yeah. |
| 15:32.48 | ManxPower | OT: I had a bunch of reports from users that the mail server was finding the "Can't" virus in their e-mails. Turns out the real message is something like "Can't open spamassassin socket". |
| 15:32.53 | *** join/#asterisk intralanman (n=lanman@pool-71-253-247-137.nrflva.east.verizon.net) |
| 15:33.01 | sp0n9e] | how do i tell which phone number i'm being called with? (we have line hunting so the phone call might come in on a different number) |
| 15:34.20 | qwertz_ | I've got some snoms and want to use the call pickup feature. I added hints to each telephone and configured a destination key for the telephone. So when tel1 calls tel2 I can see it at tel3 a blinking led. But when I press the button it doesn't pickup but starts a new call. So am I doing something wrong? |
| 15:34.58 | M_at | sp09e - How are the lines presented? |
| 15:35.08 | sp0n9e] | presented? |
| 15:35.15 | M_at | analogue, isdn etc |
| 15:35.22 | sp0n9e] | oh, analog |
| 15:35.34 | sp0n9e] | POTS through a sangoma a200 card |
| 15:36.38 | ontae | ManxPower & M_at : O.K. SIP through NAT is now clear, I think. May you can help me with my "401 Unauthorized"? May I put a "sip debug" trave somewhere? |
| 15:37.38 | *** join/#asterisk Del_Mon (n=de_mon@fl-69-69-137-201.dyn.embarqhsd.net) |
| 15:37.48 | ManxPower | unauthorized means "auth info did not match anything" |
| 15:38.03 | ManxPower | But I have to go fight the "can't" virus. |
| 15:38.07 | M_at | sp0n9e - Ask your telco if they support DDI but you may lose CLI on an analogue line. |
| 15:38.37 | PupenoR | How do I get information about dialplan functions/applications in the Asterisk console ? |
| 15:38.45 | Juggie | 'show applications' |
| 15:39.27 | PupenoR | Juggie: thanks. |
| 15:39.49 | *** join/#asterisk toxap (n=toxap@213.227.193.75) |
| 15:40.17 | PupenoR | What does it mean to dial Local/90@agent-autologin/n ? |
| 15:40.28 | *** join/#asterisk dsfr (i=dsfr@pdpc/sponsor/digium/dsfr) |
| 15:41.15 | ontae | Does anyone know, if there is something like a "Solution Database regarding asterisk problems"? |
| 15:41.26 | b11d | hahaha |
| 15:41.34 | zoa | www.asteriskguru.com has some |
| 15:41.38 | zoa | but not too much :) |
| 15:41.52 | sp0n9e] | M_at: well, the situation is funny. i've got voip to a cisco box from the telco, and i convert back to voip in the asterisk box...they didn't want to support me on straight voip |
| 15:41.57 | zoa | http://www.asteriskguru.com/tutorials/verbose_messages.html |
| 15:42.00 | b11d | PupenoR.. it means to dial channel local/90 in the agent-autologin context.. |
| 15:42.00 | zoa | add more if you want to |
| 15:42.01 | b11d | doesnt it |
| 15:42.07 | file | PupenoR: pupeno? |
| 15:42.09 | sp0n9e] | you think the cisco IAD would do DDI? |
| 15:42.15 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 15:43.44 | ManxPower | ANALOG does not provide DDI info. |
| 15:44.03 | exarverspuy | Any Digium G729 codec Guru around? |
| 15:44.18 | PupenoR | file: mh? |
| 15:44.35 | file | PupenoR: I just came across one of your bugs, can you hop into #asterisk-bugs? |
| 15:44.53 | ontae | zoa: Thanks for the info |
| 15:45.32 | *** join/#asterisk marv[work] (n=timr@br0.asteriasgi.com) |
| 15:45.39 | ontae | Another question: Is it possible to write "sip debug" to a file ? |
| 15:45.44 | PupenoR | sure. |
| 15:46.37 | *** join/#asterisk andresmujica (n=andresmu@201.244.199.222) |
| 15:48.26 | *** join/#asterisk dsfr (i=dsfr@pdpc/sponsor/digium/dsfr) |
| 15:48.46 | xnon | a Good SIP Softphone??????????????? |
| 15:49.00 | xnon | XLITE no Please! |
| 15:51.02 | *** join/#asterisk danbrwn (n=danny@216.77.58.40) |
| 15:51.59 | b11d | there ARE NONE |
| 15:52.02 | ManxPower | All Softphones Suck. |
| 15:52.03 | b11d | write one |
| 15:52.06 | b11d | and become wealthy |
| 15:52.10 | b11d | or.. poor.. |
| 15:52.14 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-42.ph.ph.cox.net) |
| 15:54.21 | M_at | ManxPower - DDI can be provided on analogue but it isn't, as far as I am aware, an international standard |
| 15:54.47 | M_at | We used to use a huge old analogue GPT pbx about 7 years ago and earlier and we had DDI on that |
| 15:56.21 | *** join/#asterisk santibiotico (n=santi@172.Red-88-1-223.dynamicIP.rima-tde.net) |
| 15:56.23 | santibiotico | hi |
| 15:56.48 | santibiotico | can anyone help me with a b410p installation, plz |
| 15:57.45 | zoa | i could i you pay me for it :p |
| 15:57.53 | zoa | try install BC |
| 15:58.10 | zoa | 99% of the questions we get about b410p is due to a lack of bc |
| 15:58.59 | santibiotico | what's bc? |
| 15:59.23 | zoa | a calculator |
| 15:59.55 | santibiotico | well i was only asking for the b410p installation procedure |
| 16:00.16 | zoa | sec |
| 16:00.19 | zoa | i have it somewhere |
| 16:00.37 | *** join/#asterisk Katty (n=angela@hera.copi-rite.com) |
| 16:00.39 | Katty | morning. |
| 16:01.37 | *** join/#asterisk nosbig (n=nosbig@rrcs-70-60-162-114.central.biz.rr.com) |
| 16:01.58 | pingwin | xnon have you tried kphone? |
| 16:02.41 | zoa | gimme your email santibiotico |
| 16:03.54 | zoa | http://www.asteriskguru.com/tutorials/digium_b410p_installation_guide.html |
| 16:04.01 | zoa | i published it just for you :p |
| 16:04.43 | docelmo | ZOA! |
| 16:04.53 | zoa | hey ho |
| 16:04.54 | docelmo | Dude how the hell are ya |
| 16:04.56 | zoa | mr ex goatie |
| 16:05.05 | docelmo | How's the crew? |
| 16:05.17 | zoa | good i think |
| 16:05.21 | zoa | did you see the pictures ? |
| 16:05.27 | docelmo | Some of them |
| 16:05.35 | docelmo | Ive been wicked busy with work |
| 16:05.41 | zoa | http://www.asteriskguru.com/gallery/main.php?g2_itemId=61 |
| 16:06.16 | RoyK | nice picture. background is in focus and all |
| 16:07.04 | b11d | nice beard.. looks like mine |
| 16:07.19 | santibiotico | zoa |
| 16:07.21 | file | ack it's docelmo |
| 16:07.31 | santibiotico | i followed the installation guide you wrote just for me :P |
| 16:07.51 | santibiotico | but i was asking because i'm getting an error message which i don't know how to solve |
| 16:08.20 | zoa | check the errors at the end of the page |
| 16:08.22 | *** join/#asterisk rsd (n=chaos@200.181.133.130) |
| 16:09.11 | docelmo | I look like I am fucking stoned |
| 16:09.32 | zoa | you probably weree |
| 16:10.36 | *** join/#asterisk tr2x (n=alvar@80-218-150-90.dclient.hispeed.ch) |
| 16:11.33 | danbrwn | trying to make the zaptel sources, where do the kernel sources need to be installed? |
| 16:13.05 | *** join/#asterisk grantm (n=grantm@207.88.78.2) |
| 16:15.30 | santibiotico | when i try to load the driver by typing /etc/init.d/misdn-init start |
| 16:15.39 | santibiotico | then i get the following error message: |
| 16:15.43 | santibiotico | FATAL: Error inserting hfcmulti (/lib/modules/2.6.17-2-amd64/extra/hfcmulti.ko): Invalid argument |
| 16:16.56 | *** join/#asterisk stephane__ (n=stephane@merlin.cabale.net) |
| 16:17.40 | santibiotico | i understanf i need to install the calculator |
| 16:18.14 | santibiotico | but i don't know what bc is...heheh |
| 16:18.31 | Qwell[] | binary calculator |
| 16:19.34 | *** part/#asterisk jaike (n=jaike@210.213.125.121) |
| 16:19.37 | santibiotico | ok i'm doing it right now |
| 16:19.38 | santibiotico | thanks |
| 16:22.29 | *** join/#asterisk E-bola (i=ice@rbii-valhalla.mrseb.co.uk) |
| 16:28.44 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 16:28.47 | *** join/#asterisk Ebola (n=Ebola@host86-136-86-71.range86-136.btcentralplus.com) |
| 16:29.07 | *** part/#asterisk Dio_ (n=dima@arkadia.soborka.net) |
| 16:31.16 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.149) |
| 16:33.20 | *** join/#asterisk merbanan (n=Anders@136.240.13.217.in-addr.dgcsystems.net) |
| 16:35.21 | *** join/#asterisk jart (n=user@ool-44c04b3a.dyn.optonline.net) |
| 16:38.35 | *** join/#asterisk alamantia (i=alamanti@nat/digium/x-79a14a957bc96697) |
| 16:39.17 | *** join/#asterisk spunz (n=spunz@h081217096236.dyn.cm.kabsi.at) |
| 16:39.48 | *** join/#asterisk hfb (n=hfb@pool-71-118-252-158.lsanca.dsl-w.verizon.net) |
| 16:40.57 | *** join/#asterisk SplasPood (n=jwb@nat-loopback.lga4.us.corp.voxel.net) |
| 16:42.37 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 16:43.23 | *** join/#asterisk SplasPood (n=jwb@nat-loopback.lga4.us.corp.voxel.net) |
| 16:44.06 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 16:46.00 | *** join/#asterisk _cleric_ (n=dacleric@p548202D8.dip0.t-ipconnect.de) |
| 16:47.49 | *** join/#asterisk beBBo (n=bebbo@85-18-14-22.fastres.net) |
| 16:48.24 | beBBo | hi all :) |
| 16:49.30 | *** join/#asterisk root (n=root@199.227.185.35) |
| 16:49.56 | beBBo | someone can help me with UUS [ISDN features] please? I'm going crazy :\ |
| 16:51.47 | *** join/#asterisk BSDTech (n=RNeese@adsl-69-230-170-5.dsl.irvnca.pacbell.net) |
| 16:53.33 | backblue | omg using misdn on amd64... |
| 16:58.10 | *** join/#asterisk fall0ut (i=tim@realfuckingnews.com) |
| 17:00.13 | ManxPower | beBBo: define UUS |
| 17:00.55 | fall0ut | so, does asterisk have working SIP-B bridged line apperances yet? |
| 17:01.10 | file | argh |
| 17:01.28 | ManxPower | fallOut I don't think so, but I could be wrong. |
| 17:01.41 | file | no it does not |
| 17:01.48 | fall0ut | weak |
| 17:01.54 | fall0ut | what about with MGCP? |
| 17:02.23 | Qwell[] | fall0ut: patches accepted. :) |
| 17:02.31 | fall0ut | wait |
| 17:03.00 | fall0ut | asterisk mgcp has no RFC3149-type stuff, either? |
| 17:03.02 | fall0ut | right? |
| 17:04.05 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:04.26 | fall0ut | How do ya'll get over people needing Key/Hybrid type functions then? |
| 17:04.39 | ManxPower | Fallout we use the non SIP-B BLF stuff in Asterisk |
| 17:04.41 | blitzrage | do the Polycom's not let you set the TFTP boot server in the .cfg files? I can't find it anywhere |
| 17:04.50 | ManxPower | blitzrage: no. |
| 17:05.03 | ManxPower | blitzrage: you can set it manually in the boot menu or via DHCP |
| 17:05.03 | blitzrage | could have sworn the old configs let you set that |
| 17:05.18 | file | blitzrage: back to work, slacker |
| 17:05.27 | blitzrage | file: I am working.... |
| 17:05.32 | *** join/#asterisk slaq (i=foobar@port-212-202-38-87.dynamic.qsc.de) |
| 17:06.21 | ManxPower | fallout: I don't know the actual protocol number, but BLF works just fine with Polycoms |
| 17:06.50 | fall0ut | Hrm... But the BLF doesnt allow you to seize lines though |
| 17:06.52 | fall0ut | right? |
| 17:06.57 | fall0ut | say they put the call on hold |
| 17:06.59 | fall0ut | and go to another phone |
| 17:07.02 | fall0ut | you cannot just pick the call up |
| 17:07.13 | *** join/#asterisk andresmujica (n=andresmu@201.244.199.222) |
| 17:07.15 | ManxPower | This is starting to piss me off. There are almost no 900Mhz DSS phones in stock at Amazon |
| 17:07.24 | ManxPower | fallout: no. We use call parking for that |
| 17:08.07 | Juggie | why do you need 900mhz? |
| 17:08.07 | backblue | blitzrage: makes no sense doing that in the .cfg |
| 17:08.08 | ManxPower | You can either fight Asterisk's oddities and be miserable and depressed or you can work with Asterisk's oddities and live a happy complete life. |
| 17:08.24 | ManxPower | Juggie: because of trees, plants, bushes, hills |
| 17:08.46 | Juggie | i guess 5.8 isnt a good answer then :) |
| 17:09.24 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
| 17:09.24 | ManxPower | The old 900Mhz DSS phone died, the NEW USA DECT 1.9 Ghz phone we tries SUCKED, 5Ghz gives us almost no range in this envioment, and 2.4Ghz would piss off the WiFi stuff |
| 17:09.29 | *** join/#asterisk andresmujica (n=andresmu@201.244.199.222) |
| 17:10.14 | blitzrage | backblue: it does when you don't have physical access to the phone, and its on someone elses network, and you want to change the boot server IP |
| 17:10.24 | *** join/#asterisk topping_ (n=topping@natip.kink.com) |
| 17:10.38 | fall0ut | hrm... So that sucks... |
| 17:10.43 | backblue | blitzrage: it needs the tftp to get the .cfg, so it does not. |
| 17:10.46 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
| 17:11.07 | backblue | fall0ut: the implementation you are looking for it's behing developed in freeswitch. |
| 17:11.17 | ManxPower | blitzrage: Let me introduce you to this new fangled thing called DNS. |
| 17:11.46 | backblue | and dhcp server should provide it, just change the dhcpd.conf |
| 17:11.46 | blitzrage | ManxPower: meh... DNS is for nerds |
| 17:11.49 | heison | has anyone here successfully loaded ztdummy in xen domU? |
| 17:11.50 | *** join/#asterisk p0g0 (n=pogo@madwifi/support/p0g0) |
| 17:11.54 | fall0ut | backblue: yea... |
| 17:12.04 | fall0ut | backblue: just kinda disappointed, wanted to use it for the office pbx, but can't... |
| 17:12.15 | backblue | fall0ut: i know, i'm too. |
| 17:12.25 | fall0ut | Sylantro/Broadsoft etc supports it |
| 17:12.32 | fall0ut | MetaSwitch does not in their SIP stack, either |
| 17:12.42 | backblue | there are some funcionality we need them on pbx software, freeswitch can ack like a pbx but it's not inteend to be it, at least now. |
| 17:12.46 | ManxPower | fallout: When you change PBXs your users will have to be retrained. No matter *which* PBX you use. |
| 17:13.03 | backblue | so, when asterisk suports |
| 17:13.06 | fall0ut | ManxPower: that's a killer for these people though |
| 17:13.22 | backblue | will just one more hack, over a couple of 1000 that it allready has. |
| 17:13.27 | ManxPower | fallout: I don't recommend that retarded people use Asterisk |
| 17:13.29 | fall0ut | And truthfully, SIP-B still sucks for it |
| 17:13.32 | backblue | but, if it works, ok for me. |
| 17:13.43 | backblue | if it's nothing more, working... |
| 17:13.46 | fall0ut | especially through SBCs |
| 17:14.16 | heison | when i tried to modprobe ztdummy in domU, it loaded zaptel and crc_ccitt, but failed to load ztdummy... it complains about unresolved symbol - rtc_register, rtc_unregister & rtc_control,turns out rtc can't load due to IRQ 8 not free (domU has no access to that...) Is there a way to build ztdummy without rtc? |
| 17:14.18 | ManxPower | I would love to see the operator console for a company with 500 lines. |
| 17:14.32 | ManxPower | heison: I don't ev en know what a domU is. |
| 17:14.34 | backblue | ManxPower: that dont exist. |
| 17:14.51 | backblue | there exists something called IVR so you can filter the calls |
| 17:14.56 | backblue | and distribute them |
| 17:15.18 | ManxPower | backblue: Stop using logic when I'm being bombastic |
| 17:15.20 | *** join/#asterisk ruskie (n=ruskie@sourcemage/mage/ruskie) |
| 17:15.25 | Katty | i hate busy days ) |
| 17:15.26 | Katty | )= |
| 17:15.36 | backblue | ManxPower: you are behing *out of reality* |
| 17:15.51 | beBBo | I'm back |
| 17:16.20 | backblue | no one, uses 500 lines consoles |
| 17:16.21 | ManxPower | backblue: Just like the people that think BLFs are the only way to handle calls. |
| 17:16.28 | beBBo | backblue: UUS is user-to-user, a ISDN feature that provide information from the provider to me |
| 17:16.35 | ManxPower | backblue: Yeah, if you have 500 lines you stop using the Fisher Price phone systems and get a real PBX |
| 17:16.39 | Katty | file: is today everything you dreamt it would be? |
| 17:16.46 | fall0ut | asterisk != real pbx |
| 17:16.47 | fall0ut | heh |
| 17:16.51 | backblue | yes |
| 17:16.55 | file | Katty: yes! |
| 17:16.59 | Katty | file: yay! |
| 17:17.03 | ManxPower | fallout: Um, you mean Asterisk != Key System |
| 17:17.11 | heison | ManxPower: Xen virtualization... |
| 17:17.14 | fall0ut | Asterisk != Real PBX |
| 17:17.16 | ManxPower | Key Systems use BLFs |
| 17:17.24 | ManxPower | PBXs do not. |
| 17:17.28 | fall0ut | Asterisk != Real Telephony Switch |
| 17:17.31 | fall0ut | it's useful |
| 17:17.38 | fall0ut | I will give it that |
| 17:17.42 | ManxPower | fallout: What specific feature does Asterisk lack that makes it not a real pbx? |
| 17:17.49 | Qwell[] | ManxPower: pretty lights |
| 17:18.12 | Katty | ManxPower: butbut, fisher price makes pretty blue colors! |
| 17:18.19 | blitzrage | ManxPower: it's not confusing enough |
| 17:18.31 | Katty | ManxPower: and nothin's quite like pastel blue in a server room. |
| 17:18.32 | Qwell[] | blitzrage: I think you win |
| 17:18.41 | backblue | 500 exten it's not problem for asterisk |
| 17:18.50 | backblue | making calls with all of that, can be. |
| 17:18.56 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 17:19.19 | *** join/#asterisk TonyM__ (n=TonyM@softins.claranet.co.uk) |
| 17:19.27 | Qwell[] | 500 calls? nah |
| 17:19.28 | backblue | asterisk it's a couple of patch's that works, and in lack of better, all use it. |
| 17:19.31 | Qwell[] | That's simple enough |
| 17:19.51 | ManxPower | Just route them all to Congestion() |
| 17:20.06 | fall0ut | asterisk doesn't scale :( |
| 17:20.20 | backblue | fall0ut: well, asterisk it self dont |
| 17:20.34 | backblue | but it's not supposed too |
| 17:20.44 | fall0ut | and thats not really it's fault |
| 17:20.49 | fall0ut | soft dsp doesn't scale |
| 17:20.52 | blitzrage | it scales if you know how to cluster |
| 17:20.55 | backblue | anything that scale the way it should, it's not done by only one software. |
| 17:21.11 | ManxPower | Falllout: I agree that Asterisk does not scale DOWN to the keysystem level very well.. |
| 17:21.16 | blitzrage | and know what to use it for, and where, and where to use other software |
| 17:21.26 | fall0ut | it does not scale large very well either |
| 17:21.29 | ManxPower | I can't really say much about scalling UP. |
| 17:21.31 | fall0ut | pls to see Sylantro/Broadsoft |
| 17:21.57 | *** join/#asterisk asdx (n=diego@200.61.236.33) |
| 17:22.11 | fall0ut | but asterisk is useful... |
| 17:22.18 | backblue | who needs scalling a pbx? |
| 17:22.31 | fall0ut | service providers |
| 17:22.31 | fall0ut | heh |
| 17:22.36 | backblue | no |
| 17:22.45 | beBBo | I'm going crazy with ISDN features in astersik, i need to read the "User-to-User Signalling, services 1, 2 and 3" (UUS) that my provider send to me for every incoming call... someone can help me please? |
| 17:22.46 | backblue | service providers use soft-switch |
| 17:22.50 | backblue | they dont care about any pbx |
| 17:22.51 | backblue | ... |
| 17:23.06 | ManxPower | Yeah, it sucks that service providers can't use Asterisk |
| 17:23.26 | ManxPower | beBBo: Asterisk does not support ISDN BRI |
| 17:23.27 | backblue | no it doesnt |
| 17:23.53 | ManxPower | If service providers could use Asterisk they might start supporting IAX2 |
| 17:24.06 | fall0ut | haha |
| 17:24.19 | fall0ut | iax for the lose |
| 17:24.37 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:24.42 | Katty | i kinda like iax. |
| 17:24.43 | ManxPower | beBBo: I assume you are using ISDN BRI with the BRIStuff patches. Have you considered asking on the BRIstuff mailinglist (do they even exist) |
| 17:25.02 | rob0 | IAX2 is warm and soft and cuddly. |
| 17:25.07 | file | Katty: it allows you to call me! |
| 17:25.11 | Katty | file: yes! |
| 17:25.11 | ManxPower | Oh wait! Asterisk does not support BRI via mISDN and the B410P or whatever it is. |
| 17:25.23 | Katty | file: which i'm not doing right now, on account of cheese pizza. |
| 17:25.26 | *** join/#asterisk awannabe (n=gti@ip24-251-135-202.ph.ph.cox.net) |
| 17:25.28 | file | Katty: :( |
| 17:25.31 | file | Katty: may I have some? |
| 17:25.43 | Katty | kay. room 110 |
| 17:25.49 | backblue | soo we need asterisk for suport IAX2? |
| 17:25.50 | rob0 | Cheese pizza is warm and soft. |
| 17:25.51 | file | okay! be there in 10 hours |
| 17:25.52 | backblue | :) |
| 17:25.55 | backblue | new one... |
| 17:26.01 | awannabe | hey guys, how can you have the default option for a auto attendant (if no option is chosen) to dial a certrain extension, or something? |
| 17:26.14 | rob0 | (Unless it's not, I guess.) |
| 17:26.16 | shido6 | t |
| 17:26.18 | fall0ut | signaling and audio should be seperate |
| 17:26.18 | shido6 | timeout |
| 17:26.34 | Katty | file: you know what i really wanna know? |
| 17:26.41 | file | Katty: my credit card information? |
| 17:26.42 | awannabe | exten => t,1,Dial(SIP/bla,20,tr) ? |
| 17:26.46 | Katty | file: no |
| 17:26.51 | file | Katty: good |
| 17:26.52 | *** join/#asterisk apardo (n=apardo@62.15.239.65) |
| 17:26.55 | shido6 | before that do you have a time set? |
| 17:26.57 | Katty | file: why, when i try to dial 110, it tries to catch it as 11 first instead. |
| 17:27.04 | file | Katty: silly Polycom phone? |
| 17:27.12 | Katty | file: possibly. |
| 17:27.17 | Katty | file: i've not looked into it yet. |
| 17:27.27 | beBBo | ManxPower: I have a digium PRI card... [sorry for my bad english] are you sudjesting me to contact the digium mailinglist? |
| 17:27.27 | *** join/#asterisk xez (n=xez@serial.trust-it.gr) |
| 17:27.47 | fall0ut | Katty: <digitmap dialplan.digitmap="x.T|*xx.T" dialplan.digitmap.timeOut="4"/> |
| 17:28.08 | Katty | fall0ut: you do not parse. |
| 17:28.12 | ManxPower | beBBo: Yes, ask on the Asterisk-Users mailing list. Very few people with Asterisk care about the low level protocol. |
| 17:28.24 | Katty | fall0ut: mind putting that in Kat? |
| 17:28.31 | ManxPower | Katty: paste your digitmap |
| 17:28.42 | beBBo | ManxPower: thank you :) |
| 17:28.58 | jart | any of you asterisk hackers need a 1-day job? |
| 17:29.00 | fall0ut | Katty: phone's config (sip.cfg) |
| 17:29.12 | fall0ut | under <dialplan> |
| 17:29.20 | fall0ut | change the digitmap |
| 17:29.28 | Katty | uno momento. |
| 17:29.29 | awannabe | shido6: thanks :) |
| 17:29.33 | Katty | i currently have pizza issues. |
| 17:29.48 | Qwell[] | Katty: 800-pizzaholics |
| 17:30.02 | Katty | Qwell[]: yes, dear. that's me. |
| 17:30.16 | beBBo | :ManxPower: do you know if is it possible monitor all what my provider send to my with applications like tcpdump? |
| 17:31.15 | *** join/#asterisk JayTee52 (n=jtforde0@c-69-137-243-25.hsd1.in.comcast.net) |
| 17:32.34 | JayTee52 | where is the number to access an outside trunk defined? i.e. I dial a 9 to get an outside line. |
| 17:33.13 | *** part/#asterisk steveaj (n=sj@62.55.147.53) |
| 17:35.04 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 17:35.33 | fall0ut | So, onto another probably pointless question... asterisk support using imap for voicemail? |
| 17:35.44 | ManxPower | beBBo: pri debug span X and pre intense debug span X |
| 17:35.54 | ManxPower | fallout: 1.4 is supposed to. |
| 17:36.10 | *** join/#asterisk Arc_Ressiv (n=ilvantus@67.108.111.130.ptr.us.xo.net) |
| 17:36.13 | ManxPower | JayTee52: This is not a freepbx/trixbos channel |
| 17:36.21 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:36.42 | fall0ut | 1.4... cool... atleast I can use it for voicemail |
| 17:36.45 | JayTee52 | ManxPower, I don't understand. |
| 17:37.21 | ManxPower | JayTee52: "trunk" is not a term Asterisk uses. You also did not say what technology (Zap/SIP/etc) you are using. |
| 17:37.39 | ManxPower | So I assumed you used the Asterisk "GUIs" |
| 17:37.42 | *** join/#asterisk ucfMethod (n=ucfmetho@c2.efb7d1.client.atlantech.net) |
| 17:37.49 | Katty | fall0ut: yay! i found it! |
| 17:38.08 | ManxPower | Pizza! It does a brain good! |
| 17:38.13 | Katty | fall0ut: tho, now, i'm not quite sure what to do with it. none of this is in Kat. |
| 17:38.23 | Katty | silly Polycom developers! |
| 17:38.28 | Katty | they should have documented their cfg file! |
| 17:38.33 | Katty | ...in the cfg file. |
| 17:38.34 | ManxPower | Katty: polycoms can be a bitch to learn |
| 17:38.40 | ManxPower | Katty: it is in the admin manual |
| 17:38.51 | Katty | gasp, read?! |
| 17:38.52 | Katty | k |
| 17:39.00 | JayTee52 | I'm using Asterisk and I'm using both Zap and SIP. I'm trying to route an outbound call out of an FXO port. I have inbound dialing working and SIP to SIP but I can't seem to get outbound going using the examples. |
| 17:39.33 | ManxPower | Katty: if you want to lean about how to use the polycom digitmap, it is modeled after the MGCP digitmap stuff, look at the MGCP RFC, that should help. |
| 17:40.01 | ManxPower | JayTee52: ignoregidit => 9 for Zap. For SIP it is set in the SIP phone. |
| 17:40.02 | fall0ut | Katty: change it to what I posted to you |
| 17:40.09 | Katty | fall0ut: yeah,butbut. |
| 17:40.14 | Katty | fall0ut: it still won't parse. |
| 17:40.23 | Katty | fall0ut: it's all fine and dandy to fix it, but i don't get the fix :P |
| 17:40.30 | fall0ut | <digitmap dialplan.digitmap="x.T|*xx.T" dialplan.digitmap.timeOut="4"/> |
| 17:40.31 | beBBo | ManxPower: do you know if is it possible see the UUI? (User-to-User Information) |
| 17:40.35 | Katty | sigh. |
| 17:40.38 | ManxPower | Also remember you do NOT "select an outside line" You dial a prefix digit (9 in the USA) and it is sent as part of the number to the dialplan, where you have to route the call based on that and strip that digit before sending the call the the telco |
| 17:40.43 | *** part/#asterisk TonyM__ (n=TonyM@softins.claranet.co.uk) |
| 17:40.44 | Katty | fall0ut: i'd rather comprehend it than just fix it (= |
| 17:41.03 | fall0ut | oh |
| 17:41.03 | ManxPower | beBBo: I don't know for sure but the pri debug stuff should dump the raw protocol |
| 17:41.03 | fall0ut | heh |
| 17:41.07 | fall0ut | see the MGCP RFC |
| 17:41.13 | Katty | fall0ut: thanks for the answer tho. |
| 17:41.16 | Katty | fall0ut: kings to you. |
| 17:41.58 | fall0ut | I've got some good examples of US dialplans that work, but what I pasted works 99.99% of the time |
| 17:42.00 | ManxPower | Katty: Just remember that the digit map digit T never actually SOLVES anything, IT just works around things. |
| 17:42.00 | fall0ut | just has a 4sec delayu |
| 17:42.24 | ManxPower | i.e. works around being too lazy to write a real dialplan. |
| 17:42.30 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 17:42.32 | ManxPower | and works around crappy national dialplans |
| 17:42.32 | JayTee52 | ManxPower, thanks, I've got that far from the docs but I'm trying to route through a standard analog line on my old Nortel PBX. I should be able to just pass the digits out of the Zap channel I'm pointing to in the outbound section of extensions.conf. |
| 17:43.03 | ManxPower | JayTee52: exten => 9,1,Dial(Zap/1/) |
| 17:43.07 | ManxPower | remember the trailing / |
| 17:43.24 | BSDTech | I say if your to lazy to write a dial plan get freepbx |
| 17:43.30 | BSDTech | and asterisk |
| 17:43.48 | *** part/#asterisk upod (n=chibondk@cpe-66-108-211-222.nyc.res.rr.com) |
| 17:44.06 | fall0ut | 0[2-9]XXXXXXT|[2-9]XXXXXXT|0[2-9]XX[2-9]XXXXXX|[2-9]XX[2-9]XXXXXX|01[2-9]XXXX.T|011[2-9]XXXX.T|101XXXX|950XXXX|[0-9*].#|0[2-9]XXXXXXXXX|1[2-9]XXXXXXXXX|[2-9]11|0[2-9]11|1[2-9]11|0T|00|555XXXX|*[014-9]X|11[23]XX|11[014-9]X|*[2-3]XX|101T|10[02-9]|958XXXX|959XXXX |
| 17:44.13 | fall0ut | is a good one for PSTN :) |
| 17:44.27 | ManxPower | dialplan.digitmap="9,1[2-9]xx[2-9]xxxxxx|9,[2-9]xxxxxx|[2-8]xxx|9,[2-9]11|911|1x|9,011x.T|*xx" |
| 17:44.29 | *** join/#asterisk kratzers (n=kratzers@martha.pa.net) |
| 17:44.56 | Katty | i don't get any of that digitmap stuff. |
| 17:45.01 | backblue | can i specify a stun server with asterisk? |
| 17:45.02 | Katty | i didn't even know it existed before today! |
| 17:45.13 | Katty | so hot dog, i learned somethin new. |
| 17:45.15 | ManxPower | Since there is NO way to know what the length of numbers are for international I use T |
| 17:45.42 | BSDTech | 011. |
| 17:45.43 | ManxPower | Katty: it is similar (at least in concept) to Asterisk's dialplan wildcards, but a bit more powerful in some ways |
| 17:45.52 | BSDTech | or 011* |
| 17:45.57 | ManxPower | BSDTech: that STILL makes you wait for digittimeout |
| 17:46.01 | BSDTech | in this case for digimap |
| 17:46.03 | fall0ut | the one I pasted covers pretty much everything |
| 17:46.09 | Katty | ManxPower: i dig me some wildcards. |
| 17:46.11 | ManxPower | The goal is to NOT wait for digittimeout |
| 17:46.24 | Katty | ManxPower: but i think i'm only using wildcards in 4 spots. |
| 17:46.26 | fall0ut | including test lines, intl, svcs, operator, n11, n+10, international, 7 * 10d |
| 17:46.35 | Katty | 3 spots. |
| 17:46.53 | Katty | xxx, xxxxxxx, and 1xxxxxxxxxx |
| 17:47.05 | kratzers | is it possible to execute an application when calls are transferred? |
| 17:47.13 | Katty | this is like a porn flick! |
| 17:47.22 | ManxPower | kratzers: that depends on the type of transfer |
| 17:47.35 | BSDTech | I need a script that converts extensiions.conf into extensions.ael |
| 17:47.55 | ManxPower | kratzers: for device native transfers, I don't think so. At least not the way you think. |
| 17:48.05 | BSDTech | is there such a script yet |
| 17:48.14 | ManxPower | for DTMF Hack Transfers (see features.conf) you should be able to. |
| 17:48.24 | kratzers | ManxPower: thanks |
| 17:48.34 | JayTee52 | ManxPower, thank you very much. Gotta run back to work and try that. |
| 17:48.50 | ucfMethod | does anyone know how to test 911? |
| 17:48.52 | kratzers | need to record most every call, and I'd like to record to new files when a call is transferred |
| 17:49.04 | ucfMethod | without actually calling them, i want to make sure they can see our address etc. |
| 17:49.06 | *** join/#asterisk angryuser (n=Miranda@i03v-213-44-169-43.d4.club-internet.fr) |
| 17:49.18 | angryuser | hi everybody |
| 17:49.35 | BSDTech | who is your e911 provider |
| 17:49.40 | angryuser | anybody use astribank here? |
| 17:49.45 | ManxPower | ucfMethod: route 911 to a playback application. Once that is done and tested then send the calls to real 911 outside of busy times and ALWAYS talk to the 911 operator to tell them this is a test of the company PBX 911 dialing |
| 17:49.49 | ucfMethod | i already have the dialmaps setup in the polycom phones, and i know it will work if i actually dial 911, but i dont think they will be happy with "oh this is just a test of my voip system" |
| 17:50.05 | ManxPower | confirm the calls are seen by the 911 PSAP as having the right numbers. |
| 17:50.21 | ucfMethod | BSDTech: Vitelity |
| 17:50.39 | ManxPower | ucfMethod: So you are not using Zap? Then I have no advice. |
| 17:51.22 | ucfMethod | ManxPower: Nope, using IAX2 to Vitelity for PSTN termination |
| 17:51.48 | ucfMethod | ManxPower: everything works beautifully |
| 17:52.31 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 17:55.57 | *** join/#asterisk p0g0 (n=pogo@madwifi/support/p0g0) |
| 17:57.12 | *** join/#asterisk brian (i=brian@unaffiliated/brian) |
| 17:57.21 | brian | hey i'm messing around with asterisk on freebsd |
| 17:57.42 | brian | i have a basic dial plan setup that sends calls to the DID to a meetme conference |
| 17:58.08 | brian | When you are a the first caller when it's supposed to say "You are currently the only person in this conference" |
| 17:58.18 | brian | Instead it just says "currently the only person in this conference" |
| 17:58.30 | brian | How do I fix that? |
| 17:58.45 | Juggie | put an Answer() before Meetme() |
| 17:59.01 | ucfMethod | Wait(1) |
| 17:59.14 | ucfMethod | also works nicely. you wont even notice it |
| 17:59.50 | ManxPower | brian: Frequently it takes a few moments to set up the RTP audio, so a wait after an Answer will help. I usually use a Wait(1). |
| 17:59.52 | brian | also when you press a DTMF I noticed it makes a really nasty noise |
| 18:00.03 | *** join/#asterisk jm|home (n=jamiem@zen.jamiem.com) |
| 18:00.03 | brian | How do I make it mute out the DTMF |
| 18:00.08 | ManxPower | Just doing an Answer won't help because the call is ALREADY being answered by MeetMe |
| 18:00.35 | ManxPower | brian: what makes you so sure asterisk is detecting the tones as DTMF? |
| 18:01.30 | brian | I'm using ulaw codec...What codec should I be using? |
| 18:01.56 | Juggie | ManxPower, you assume Meetme does things in the proper order :) |
| 18:02.07 | Juggie | it may start to play the audio before it does the answer. |
| 18:02.17 | ManxPower | brian: You use whatever DTMF mode your PROVIDER is using. |
| 18:02.35 | *** join/#asterisk SupeR (n=_-Sarah-@85.106.174.102) |
| 18:02.43 | ManxPower | Juggie: it could, but I AM assuming it works similar to most of the other Asterisk applications that do answer by default before playing audio |
| 18:03.15 | *** join/#asterisk asberger (n=nb@cpe-76-170-93-79.socal.res.rr.com) |
| 18:03.23 | brian | ManxPower: Answer() and Wait(1) both don't help |
| 18:03.56 | ManxPower | brian: does Wait(5) help? If so maybe you need a bit more than 1 |
| 18:08.26 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 18:09.14 | *** join/#asterisk ShipHead (i=ShipHead@gateway/tor/x-9debf4aa19819cd8) |
| 18:09.46 | *** join/#asterisk Mattwj2005 (n=Matt@c-76-17-131-68.hsd1.mn.comcast.net) |
| 18:09.52 | Mattwj2005 | hey guys :) |
| 18:10.02 | Mattwj2005 | quick question for you.....what is the best tos? |
| 18:10.32 | Mattwj2005 | good afternoon by the way |
| 18:10.39 | monsted | Mattwj2005: well, ST:TOS isn't very good |
| 18:10.42 | codefreeze | BSDTech: no such script yet (but it's been requested as an enhancement). |
| 18:10.57 | Mattwj2005 | haha monsted |
| 18:11.07 | Mattwj2005 | are your more of a Next Gen type of guy? |
| 18:11.12 | *** join/#asterisk Math` (n=privmath@bas4-montreal19-1242358449.dsl.bell.ca) |
| 18:11.43 | monsted | Mattwj2005: TNG and Voyager, yeah |
| 18:11.53 | Mattwj2005 | yeah Voyager is my favorite |
| 18:12.00 | Mattwj2005 | six of nine rocks! |
| 18:12.06 | Mattwj2005 | *seven |
| 18:13.12 | Mattwj2005 | lol nice |
| 18:13.37 | ManxPower | Voyager was pretty cool. |
| 18:13.37 | rob0 | (transmission was about to give out) |
| 18:13.46 | Mattwj2005 | well I would need is a halodeck and a replecator and I would be set for life :) |
| 18:13.46 | angryuser | i am unable to compile zaptel/xpp/utils for astribank under feroda core 5 i got this http://pastebin.ca/259703 any ideas? |
| 18:13.51 | ManxPower | Kicking Borg Ass, and 7of9's ass. |
| 18:14.59 | *** join/#asterisk cbm11211 (n=AB@66.250.98.174) |
| 18:15.57 | Mattwj2005 | seriously though |
| 18:16.03 | Mattwj2005 | what is the best value for tos? |
| 18:16.10 | Qwell[] | Mattwj2005: 42 |
| 18:16.13 | monsted | i don't think i'd be *kicking* 7of9s ass |
| 18:16.15 | *** join/#asterisk WAudette (n=WAudette@c-71-237-146-239.hsd1.or.comcast.net) |
| 18:16.18 | monsted | Mattwj2005: EF? |
| 18:16.22 | Qwell[] | monsted: yeah, seriously |
| 18:16.29 | ManxPower | ARGH! Uniden has a "phone finder" page. I select the ONLY requirement as being 900Mhz Digital and they have no phones with that feature. My question is "Why have the option if you don't make the product" That's about as useful as Microsoft having a "Stable Windows Version" |
| 18:16.36 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 18:16.41 | ManxPower | Mattwj2005: Letm e look |
| 18:16.47 | Qwell[] | ManxPower: Hey, that isn't fair |
| 18:16.55 | Qwell[] | ManxPower: 3.11 was quite stable |
| 18:17.05 | ManxPower | Mattwj2005: tos=0xb8 |
| 18:17.11 | Qwell[] | I mean...in comparison |
| 18:17.34 | Mattwj2005 | okay |
| 18:17.37 | ManxPower | And vtech's web site is totally broken, any useful page is "not found" |
| 18:17.49 | Mattwj2005 | so higher is better I take it :) |
| 18:18.16 | angryuser | so i will get rid if astribank then;( |
| 18:18.37 | ManxPower | Mattwj2005: NO! |
| 18:18.56 | Mattwj2005 | ok? |
| 18:19.32 | ManxPower | Mattwj2005: each code has a meaning. 0xb8 should be the same as Cisco's DSCP code EF, which is "real time" and it should also match the standard IP low latency class |
| 18:19.45 | Juggie | ManxPower, http://www.futureshop.ca/catalog/proddetail.asp?logon=&langid=EN&sku_id=0665000FS10080509&catid=22177 |
| 18:20.15 | Juggie | only one they have, and only 4 left, buy them all :) |
| 18:20.24 | Mattwj2005 | okay yeah that is what I had it set for before |
| 18:20.27 | ManxPower | Juggie: the specs refer to "channels" and that usually means "not DSS" |
| 18:20.32 | Mattwj2005 | 0x10 low latency in other words |
| 18:20.43 | *** join/#asterisk florz_ (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 18:21.01 | ManxPower | Mattwj2005: remember that every router between they two devices must support and honour the setting if you expect it to be 100% perfect |
| 18:21.29 | Mattwj2005 | yeah with my system there is no way of determining that |
| 18:21.39 | Mattwj2005 | here is my basic setup |
| 18:21.56 | Mattwj2005 | sip phone -> asterisk box -> cable modem -> WAN cloud |
| 18:22.11 | Qwell[] | public internet? yeah, you aren't gonna get QoS |
| 18:22.12 | Mattwj2005 | asterisk box is running iax |
| 18:22.30 | Mattwj2005 | I have had pretty good luck with 0x10 |
| 18:22.39 | Mattwj2005 | for the tos |
| 18:22.46 | Juggie | ManxPower, i've been looking aronud a bit, and all i am finding is 900mhz analog. |
| 18:23.04 | Mattwj2005 | it is pretty usable 6+ Mbps down 384 kbps up |
| 18:23.12 | ManxPower | Juggie: Those are the most common. But 900Mhz does exist. The phone that died was. |
| 18:23.16 | *** join/#asterisk tegioz (n=tegioz@212.166.247.154) |
| 18:23.18 | ManxPower | ..er...900Mhz DSS |
| 18:23.32 | Juggie | you are probally going to have to ebay it |
| 18:24.00 | Mattwj2005 | I should upgrade to the 8 Mbps service....but I am trying to save money :) |
| 18:24.03 | ManxPower | Juggie: We also tried a DECT 6.0 phone. I had high hopes. |
| 18:24.09 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 18:24.25 | Mattwj2005 | that has a 768 kps up stream |
| 18:24.36 | ManxPower | Mattwj2005: it would work with .25Mbps. The size of your pipe does not matter much for 1 or two calls. |
| 18:25.08 | Mattwj2005 | oh I know |
| 18:25.10 | Juggie | ManxPower, http://froogle.google.com/froogle?q=900mhz+cordless+digital&btnG=Search+Froogle |
| 18:25.22 | toerkeium | guys, when I start recording a conversation with *1, whatever I say, the answer from the called person is "printed" over my voice. understand what I mean? lets say I call you, and ask you "how are you?", and you say fine. when I check the recording, the word "fine" from you, is printed before "how are you" ends. Any idea? |
| 18:26.14 | Mattwj2005 | so ManxPower I should try tos=0xb8 for my setup? |
| 18:26.14 | Juggie | example: http://www.ekitchengadgets.com/sonsp90dssco.html?ManufacturerId=052-SPP-SS960&002=21 |
| 18:27.37 | nortex | I need some ideas of what to look for when analog lines fail to recieve the callerid. I floated the question around last week and got some suggestions, but want to see if anyone has more ideas. The situation is the 3 line from the telco have callerid service and only one recieves it correctly the other 2 give a checksum error for the incoming call. |
| 18:29.34 | nortex | Mattwj2005, I think what you will find is that sending the call across the internet is not going to have any benefit of tos. You may get some routers out there that honor it, but most will ignore the tos in the headers and pass the packet as any other packet of data. |
| 18:30.01 | *** join/#asterisk dasenjo (n=dasenjo@63.245.86.186) |
| 18:30.04 | Mattwj2005 | okay sounds good |
| 18:30.46 | Mattwj2005 | fresh my memory.....what if the first router doesn't it honor it.....is there a possibility any of the further hops will? |
| 18:30.47 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 18:30.58 | Mattwj2005 | do they strip that information out between hops? |
| 18:31.37 | Juggie | no chance. |
| 18:31.45 | Juggie | each hop has to honor |
| 18:32.37 | Mattwj2005 | okay thanks.....I am just starting my career in networking....a lot to learn....I am trying to finish up my CCNP :) |
| 18:32.40 | RoyK | Mattwj2005: just use a jitterbuffer |
| 18:33.09 | Mattwj2005 | jitterbuffer has been no help....as matter of fact they make it worse often times |
| 18:33.19 | RoyK | plc helps |
| 18:33.20 | RoyK | and jb |
| 18:33.23 | RoyK | beleive me |
| 18:33.25 | toerkeium | what is a good g729 supported softphone? |
| 18:33.28 | RoyK | Mattwj2005: what protocols? |
| 18:33.32 | Mattwj2005 | jb? |
| 18:33.36 | RoyK | jitterbuffer |
| 18:33.51 | Mattwj2005 | well I was just using jitterbuffer on sip |
| 18:33.59 | RoyK | with 1.4? |
| 18:34.26 | Mattwj2005 | I am still using 1.2.13 |
| 18:34.35 | RoyK | the patch from asterisk-addons? |
| 18:34.46 | *** join/#asterisk asberger (n=nb@cpe-76-170-93-79.socal.res.rr.com) |
| 18:34.58 | toerkeium | jitterbuffer is a softphone?? |
| 18:35.00 | Mattwj2005 | I haven't added that yet |
| 18:35.00 | RoyK | Mattwj2005: there's no sip jb in * 1.2 without that |
| 18:35.10 | RoyK | toerkeium: most softphones have it |
| 18:35.42 | RoyK | also, I meant the one from http://asterisk-backport.org |
| 18:35.46 | Mattwj2005 | is 1.4 stable yet? |
| 18:35.55 | RoyK | is the pope buddhist? |
| 18:35.58 | *** join/#asterisk ShipHead (i=ShipHead@gateway/tor/x-1c3b1c3824e2e9a3) |
| 18:36.30 | RoyK | the jb code from asterisk-backports.org is the one that went into 1.4, and it works, we're running it in a pretty large setup in production |
| 18:36.37 | Mattwj2005 | some Christian views and Buddhist views are similar :) |
| 18:37.05 | RoyK | well, some views are similar anywhere on the planet |
| 18:37.16 | RoyK | but 1.4 is not stable and 1.2 does not have a sip jb |
| 18:37.18 | RoyK | nor plc |
| 18:37.43 | Mattwj2005 | good poing royk :) |
| 18:38.30 | *** join/#asterisk sevard (n=sev@c-67-188-173-23.hsd1.ca.comcast.net) |
| 18:39.11 | xnon | hello friends hey i need to know if anybody have a command line to know what version of asterisk is? |
| 18:39.21 | sevard | asterisk -rx 'show version' |
| 18:39.22 | RoyK | show version |
| 18:39.33 | xnon | asterisk show version? |
| 18:39.34 | Mattwj2005 | now if everyone could just get along we would be set |
| 18:39.40 | RoyK | or merely start asterisk with -v and it'll show you |
| 18:39.44 | RoyK | erm |
| 18:39.47 | RoyK | -vc |
| 18:39.50 | Mattwj2005 | asterisk -V |
| 18:40.00 | RoyK | kill -9 -1 |
| 18:40.33 | Aurs | cat asterisksource/.version |
| 18:40.42 | *** join/#asterisk mw46 (n=marco@64-142-40-224.dsl.static.sonic.net) |
| 18:41.36 | Mattwj2005 | asterisk -V doesn't start asterisk....it just shows the version |
| 18:42.07 | ManxPower | RoyK: neither has qualify smoothing in SIP |
| 18:42.30 | RoyK | what do you mean, qualify smoothing? |
| 18:44.02 | Skarmeth | hi all |
| 18:44.27 | Skarmeth | any extension monitoring software other then FOP and HUD? |
| 18:44.39 | Skarmeth | For GNU/Linux |
| 18:45.55 | Aurs | Skarmeth: gastman - The Graphical Asterisk call manager |
| 18:48.07 | *** part/#asterisk BSDTech (n=RNeese@adsl-69-230-170-5.dsl.irvnca.pacbell.net) |
| 18:50.24 | *** join/#asterisk crochat (i=crochat@84-74-145-139.dclient.hispeed.ch) |
| 18:56.20 | robin_z | hmmm ... still not got callerid working with chan_mISDN |
| 18:59.55 | ManxPower | robin_z: Did you try a Wait(1) at the beginning of the dialplan. I don't know about BRI, but on PRI the CallerID can arrive after the call setup |
| 19:00.09 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:01.38 | brian | conf_run: Unable to write frame to channel: No such file or directory |
| 19:01.44 | brian | What does that warning mean? |
| 19:04.32 | ManxPower | brian: my first guess is that the caller hungup |
| 19:04.37 | *** join/#asterisk paolob (n=donpaolo@196.3.84.214) |
| 19:04.49 | *** part/#asterisk paolob (n=donpaolo@196.3.84.214) |
| 19:04.52 | *** join/#asterisk dasenjo (n=dasenjo@201.228.128.10) |
| 19:05.05 | *** part/#asterisk henrique (n=henrique@201-1-131-176.dsl.telesp.net.br) |
| 19:05.10 | *** join/#asterisk p0g0 (n=pogo@madwifi/support/p0g0) |
| 19:05.46 | *** part/#asterisk Mattwj2005 (n=Matt@c-76-17-131-68.hsd1.mn.comcast.net) |
| 19:07.03 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 19:08.42 | *** join/#asterisk Isaiah (n=Isaiah@70-101-148-224.dr01.hnvr.mi.frontiernet.net) |
| 19:10.06 | *** join/#asterisk BSDTech (n=RNeese@adsl-69-230-170-5.dsl.irvnca.pacbell.net) |
| 19:10.49 | BSDTech | is there plans at anytime oin the future other then the asterisk-bsd group to make bsd a supported os for asterisk ? |
| 19:11.08 | BSDTech | or are porters just spinning thier wheels |
| 19:11.23 | Qwell[] | we already suppose bsd, to an extent |
| 19:11.26 | Qwell[] | support too |
| 19:11.41 | *** join/#asterisk lullabud (n=lullabud@12.24.42.67) |
| 19:11.56 | BSDTech | yes its in ports but does digium ever plan to support it or just those here ? |
| 19:12.07 | Qwell[] | depends on your definition of support |
| 19:12.17 | BSDTech | like for hardware |
| 19:12.25 | Qwell[] | highly unlikely |
| 19:12.54 | Qwell[] | the zaptel drivers would be completely different between OSs |
| 19:12.54 | BSDTech | ok |
| 19:13.19 | BSDTech | we have a port of the zaptel drivers and then 1.4 are being ported now |
| 19:13.32 | BSDTech | so thats not a issue |
| 19:13.33 | FuriousGeorge | imo, sip registering should be a p2p type deal. iow, my asterisk server should be able to subscribe to another asterisk servers subscriptions |
| 19:13.38 | FuriousGeorge | * is already domain aware |
| 19:14.27 | BSDTech | I know they have g729 for bsd |
| 19:14.30 | Qwell[] | FuriousGeorge: there are already a bunch of ways to go about that.. either using something like SER, or dundi, or something |
| 19:14.35 | BSDTech | so thats nice ofthem |
| 19:15.11 | Juggie | digium has g729 for alot of os'es |
| 19:15.17 | Juggie | much more then just linux |
| 19:15.36 | FuriousGeorge | Qwell[]: you think dundi would work for presence? ive been playing with SER for this, but its a steep learning curve and time is short |
| 19:15.47 | Qwell[] | dunno |
| 19:15.53 | Juggie | dundi doesnt really do presence |
| 19:16.06 | FuriousGeorge | so ser it is |
| 19:16.25 | Juggie | i'm no expert though, ask blitzrage |
| 19:16.41 | FuriousGeorge | i was just thinking my life would be easier if * did it, but im sure people who want p2p presence on their pbx are pretty niche atm |
| 19:18.00 | *** join/#asterisk droops (n=droops@adsl-074-245-001-031.sip.jan.bellsouth.net) |
| 19:18.12 | Juggie | explain the application. |
| 19:19.26 | Juggie | c |
| 19:19.29 | FuriousGeorge | who me? Juggie: i just want people on asterisk_server_a to know about the presence status of users of asterisk_server_b, and vice versa. i guess you could say share a roster in jabberspeek, or maybe |
| 19:19.34 | FuriousGeorge | p2p simple |
| 19:21.20 | *** join/#asterisk ESCulapio__ (n=ESCulapi@200.88.44.66) |
| 19:22.15 | ESCulapio__ | quien habla espanol |
| 19:22.35 | FuriousGeorge | yo no |
| 19:23.28 | *** join/#asterisk Cinen (n=Cinen@208.70.20.33) |
| 19:30.47 | *** join/#asterisk Nugget (i=nugget@dazed.slacker.com) |
| 19:33.40 | file | Nugget: not a slacker! |
| 19:35.08 | FuriousGeorge | can someone who knows a little bit of spanish help my man ESCulapio__ with interfacing asterisk to a cisco device via h.323. i got the language part down but h.323 is like greek to me (pun). something about registering with the call manager |
| 19:35.11 | Nugget | only for another 24 hours or so. |
| 19:35.43 | FuriousGeorge | i can translate a few phrases if you dont know spanish but think its easy but i gotta run |
| 19:35.53 | rob0 | <== slacker |
| 19:40.13 | Nugget | I'm transferring the domain this afternoon. |
| 19:41.05 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:47.59 | *** join/#asterisk marbahlarbs (n=marbahla@74.132.136.242) |
| 19:49.51 | *** join/#asterisk SirThomas (n=tomc@64.122.217.130) |
| 19:50.39 | marbahlarbs | I'm trying to talk to my pots line with an asterisk server and an fxo card. Incoming calls route correctly to the sip phone, but outbound calls don't. I tried defining a number specifically and still no-go. exten => 9381201,1,Dial(Zap/2/9381201,15) |
| 19:50.43 | marbahlarbs | what am I probably missing? |
| 19:51.56 | wunderkin | that wont matter... try adding some w before the number it dials... Zap/2/www5551212... |
| 19:52.12 | wunderkin | if you have multiple fxo ports you will want to use channel groups too |
| 19:57.09 | marbahlarbs | the www did the trick |
| 19:57.13 | marbahlarbs | you're awesome |
| 19:57.48 | brian | Is the zaptel PCI card better than the ztdummy module? |
| 19:58.35 | shimi | brian, hmm, you're comparing something dummy with something real? |
| 19:59.12 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 19:59.13 | brian | yes! |
| 19:59.25 | brian | isn't a zaptel card for hooking into a t1 line |
| 19:59.38 | ManxPower | brian: yes, but it also provides a hardware timer |
| 19:59.43 | shimi | you can't do anything with ztdummy. it's dummy |
| 19:59.51 | ManxPower | well zaptel supports many types of interfaces, not just T-1 |
| 20:02.42 | *** join/#asterisk Isaiah (n=Isaiah@70-101-148-224.dr01.hnvr.mi.frontiernet.net) |
| 20:02.43 | *** join/#asterisk ToTo (n=ToTo@host97-157-dynamic.2-87-r.retail.telecomitalia.it) |
| 20:02.47 | *** join/#asterisk ShipHead (i=ShipHead@gateway/tor/x-cf3451dd6840e26c) |
| 20:03.49 | *** join/#asterisk websae (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 20:04.00 | brian | can not having a zaptel card cause problems |
| 20:04.36 | FuriousGeorge | no, but meetme wont work w/o zaptel driver which doesnt require zaptel hw |
| 20:04.44 | FuriousGeorge | unless this has changed of course |
| 20:04.53 | ManxPower | Zaptel timing is required for MeetMe and IAX2 trunking. It can make MoH sound nicer. |
| 20:05.35 | ManxPower | ztdummy will provide the required timing, but may not be as accurate and I guess that could cause issues. |
| 20:10.08 | *** join/#asterisk Fibersrv (n=IceChat7@66-189-233-17.dhcp.cpgr.mo.charter.com) |
| 20:10.41 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 20:10.48 | *** join/#asterisk danbrwn (n=danny@216.77.58.40) |
| 20:10.52 | Fibersrv | I have a question, my company is looking into setting up an Asterisk phone system. What hardware would I need, where would I start to look? |
| 20:11.14 | danbrwn | Fibersrv: on the digium website |
| 20:12.09 | websae | anyone know what happened to TKD Fender these days? |
| 20:12.16 | Fibersrv | hmm ok, that will give a run down on the basics? I am kind of Asterisk illeiterate? <lol> I am getting my feet wet also by go ing down this avenue. |
| 20:12.53 | bkw__ | its way more than timing for meetme folks |
| 20:13.03 | bkw__ | just having the timer doesn't mean you'll get meetme to work |
| 20:13.13 | bkw__ | they use zaptel to do the muxing and stuff... its more than a clock source |
| 20:14.34 | danbrwn | Fibersrv: I am as well, there are lots of people who sell user friendly systems. I am looking for something a little more custom so am going down the hard road. It depends on your needs, but a canned system is sometimes very nice to buy |
| 20:14.35 | ManxPower | ~book |
| 20:14.38 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 20:14.39 | ManxPower | ~docs |
| 20:14.41 | jbot | from memory, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 20:14.43 | ManxPower | ~mailinglist |
| 20:14.45 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 20:15.17 | Qwell[] | macintosh asterisk mailing list? wtf? |
| 20:16.23 | Fibersrv | danbrwn, I understand the feeling, believe me. I need something to present. I appreciate the information, I will definately check it out. |
| 20:19.02 | *** join/#asterisk alerios (n=alerios@201.244.173.92) |
| 20:19.04 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 20:20.32 | *** join/#asterisk Manfish (n=themanfi@82-68-173-121.dsl.in-addr.zen.co.uk) |
| 20:25.43 | bkw__ | so so sad |
| 20:26.24 | xheliox | Hmm? |
| 20:28.00 | *** join/#asterisk ruskie (n=ruskie@sourcemage/mage/ruskie) |
| 20:28.17 | *** join/#asterisk alex_112 (n=admin@c213-89-56-72.bredband.comhem.se) |
| 20:28.38 | Manfish | what is sad? |
| 20:28.51 | ucfMethod | anyone have any pointers on this one "res_odbc.c:565 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified" I have check res_odbc.conf and res_mysql.conf |
| 20:29.14 | ucfMethod | i know its probably something real simple, but i must be missing it |
| 20:30.22 | *** join/#asterisk BSD_Tech (n=RNeese@adsl-69-230-170-5.dsl.irvnca.pacbell.net) |
| 20:30.48 | *** join/#asterisk ZaVoid (n=colin@65.244.210.44) |
| 20:31.12 | ZaVoid | hi all.. can someone point me in the right direction for wrting CDR's from asterisk to a radius server? |
| 20:31.24 | ZaVoid | having some trouble finding anythign about doing that |
| 20:32.00 | Manfish | mmmm |
| 20:32.12 | ZaVoid | hi ManxPower |
| 20:32.14 | ZaVoid | err manfish |
| 20:33.21 | Manfish | evening |
| 20:33.37 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 20:38.41 | *** join/#asterisk Sedorox_ (n=brandon@smartserv.ipv6.smart-serv.net) |
| 20:39.38 | danbrwn | if i use asterisk and non digium cards do I always have to run the ztdummy module for timing |
| 20:39.58 | ManxPower | danbrwn: do the cards have zaptel compatable drivers? |
| 20:40.04 | *** join/#asterisk purplebob (n=drussell@209.94.54.14) |
| 20:40.09 | *** part/#asterisk ZaVoid (n=colin@65.244.210.44) |
| 20:40.45 | danbrwn | ManxPower: unsure, just know that other hardware is available and trying to figure out what will be best. |
| 20:41.53 | ManxPower | danbrwn: if the card has zaptel compatible drivers, then you should not need ztdummy |
| 20:42.31 | danbrwn | ManxPower: thnks |
| 20:42.41 | ManxPower | Also remember that other brands of cards are not commonly used with Asterisk and so not many people will be able to help you. |
| 20:42.54 | ManxPower | Sangoma seems to be the most popular non-Digium card used with Asterisk |
| 20:42.59 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 20:44.14 | ManxPower | Juggie: I found the perfect phone for my needs -- except for 1 thing |
| 20:44.17 | ManxPower | price |
| 20:44.29 | ManxPower | $500 for a cordles phone is a bit much |
| 20:44.33 | danbrwn | ManxPower: yeah, the Sangoma cards are the alternate cards I was refering to. |
| 20:45.29 | *** join/#asterisk Sonderblade (n=mah@c-a558e353.131-1-64736c10.cust.bredbandsbolaget.se) |
| 20:50.48 | *** join/#asterisk santiago (n=santiago@208.195.214.146) |
| 20:55.01 | *** join/#asterisk slayer192 (n=slayer19@66.138.39.225) |
| 20:56.16 | monsted | ManxPower: cheap |
| 21:00.40 | *** join/#asterisk CleanerX (n=nix@p54A391F2.dip0.t-ipconnect.de) |
| 21:01.31 | *** join/#asterisk _santiago_ (n=santiago@debian/developer/santiago) |
| 21:01.36 | *** join/#asterisk remmo (n=chatzill@smack.isp.net.au) |
| 21:03.25 | *** join/#asterisk mivck (i=1000@ip-70-228.telesat.com.co) |
| 21:04.49 | *** join/#asterisk Mad_guy (n=austin@ip68-1-213-212.dl.dl.cox.net) |
| 21:06.04 | *** part/#asterisk lullabud (n=lullabud@12.24.42.67) |
| 21:12.34 | *** join/#asterisk jm|home (n=jamiem@dilbert.jamiem.com) |
| 21:18.20 | ESCulapio__ | Hola Quien Habla espanol |
| 21:18.46 | *** join/#asterisk ast_freak (n=jesse@h69-130-167-110.69-130.unk.tds.net) |
| 21:22.12 | ESCulapio__ | ? |
| 21:28.10 | *** join/#asterisk seele_ (n=seele@208.35.117.246) |
| 21:28.41 | seele_ | hello, the rhino channel banck can support E1 trunk or only T1 ???? |
| 21:29.18 | seele_ | rhino support E1 trunk??? |
| 21:29.42 | *** join/#asterisk [hC] (n=hardcore@S010600404521309c.vc.shawcable.net) |
| 21:31.07 | ManxPower | seele_: call Rhino. |
| 21:31.16 | ManxPower | Most channel banks come in T-1 and E-1 models. |
| 21:31.37 | seele_ | Manfish, yes like digium TDM |
| 21:31.50 | seele_ | but rhino only shows T1 |
| 21:32.30 | ManxPower | then that is all they support. |
| 21:32.34 | seele_ | ManxPower, any other economic solution for multiple FXS ? |
| 21:32.56 | ManxPower | seele_: yes, use T-1 for your Asterisk -> channel bank interface. |
| 21:33.21 | ManxPower | As long as it does not connect to the telco, you don't have to use E-1 |
| 21:33.50 | seele_ | what channel bank can I use ? |
| 21:34.52 | ManxPower | I use Adtran |
| 21:35.22 | ManxPower | so, Telco <-> E-1 <-> Asterisk <-> T-1 <-> Channel Bank <-> Analog phones |
| 21:37.00 | seele_ | aaaa |
| 21:37.44 | seele_ | Telco <-> E-1 <-> asterisk <-> Network <-> Channel Bank <-> Analog Phones |
| 21:37.52 | seele_ | si possible ? |
| 21:41.53 | *** join/#asterisk Ebola (n=Ebola@host86-136-86-71.range86-136.btcentralplus.com) |
| 21:44.03 | *** join/#asterisk ast_freak (n=jesse@h69-130-167-110.69-130.unk.tds.net) |
| 21:45.17 | *** join/#asterisk knarfly (n=maxfiles@c-65-34-177-3.hsd1.fl.comcast.net) |
| 21:48.41 | *** join/#asterisk bluregard (n=matt@67.163.72.68) |
| 21:48.47 | bluregard | hi all |
| 21:50.26 | *** join/#asterisk Katty (n=angela@hera.copi-rite.com) |
| 21:52.13 | file | Katty: haylo |
| 21:53.23 | *** join/#asterisk sloth_ (n=josh@mail.fex.org) |
| 21:53.49 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 21:54.15 | file | [TK]D-Fender: home... home on the range |
| 21:54.26 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 21:55.05 | Katty | file: ewwo. |
| 21:55.11 | sloth_ | hello i am DESPERATELY trying to get my SPA 922 to ring my custom ringtone with __ALERT_INFO. Has anyone succeeded? |
| 21:55.14 | Katty | file: i think i got my flashy time problem figured out. |
| 21:55.19 | [TK]D-Fender | Katty : Mew. |
| 21:55.24 | Katty | [TK]D-Fender: mew. |
| 21:55.34 | file | Katty: flashy time is good time though |
| 21:55.42 | [TK]D-Fender | Katty : Polycom time flashing? That'd be a SNTP issue |
| 21:55.53 | [TK]D-Fender | file : Depends on the flasher ;) |
| 21:56.03 | hmmhesays | ice ice baby |
| 21:58.11 | Katty | [TK]D-Fender: aye. |
| 21:58.21 | Katty | [TK]D-Fender: but...that's not /quite/ the issue. |
| 21:58.34 | Katty | [TK]D-Fender: i think reading sip.cfg, or maybe sip.cfg has the wrong issues. |
| 21:58.42 | Katty | [TK]D-Fender: i'm tryin some different stuff. |
| 21:58.43 | [TK]D-Fender | Katty : I'd personally suggest you just shove "pool.ntp.org" into your sp.cfg and have it override DHCP's assignment. |
| 21:59.26 | [TK]D-Fender | Katty : There are 2 other fields to indicate when SIP.CFG overrides DHCP to look for in the same clause. pretty quick to fix |
| 21:59.47 | Katty | sec, let me pastebin some stuff. |
| 22:00.04 | *** join/#asterisk CpuID2 (i=rbijrxiz@dsl-58-6-116-59.qld.westnet.com.au) |
| 22:05.31 | wunderkin | [TK]D-Fender, i'm not sure if i've asked previously.... but would you have any recommendations on things to check for an intermittant reboot problem on an ip430? it has happened with 1.6.7, 2.0.1, and 2.0.2... there does not seem to be any rhyme or reason, just talking to one person, then get one way audio.. the person im calling can hear me but i cant hear them... if they hang up, it reboots.. or if i hang up.. it reboots... |
| 22:05.57 | wunderkin | i noticed that it never completely downloads the sip.ld file but the log says that the file is fine.... and the xml editor says that the files are 'well formed' or something like that |
| 22:06.06 | *** join/#asterisk Chris-NB (n=chris@argos.campus-sbg.at) |
| 22:06.26 | [TK]D-Fender | wunderkin : nope, never seen |
| 22:06.27 | *** join/#asterisk costalivan (n=ivan@189.145.28.195) |
| 22:06.42 | costalivan | hello budies |
| 22:06.45 | ucfMethod | does anyone know if Asterisk comes with gsm files for Monday, Tuesday, Wednesday etc |
| 22:06.56 | costalivan | I'm getting some problems with siproxd software |
| 22:06.57 | ucfMethod | i searched the sound folder, and the digits folder, and found nothing |
| 22:07.25 | wunderkin | ive never heard anyone having this kind of problem with a polycom.... i tried submitting the logs to voipsupply on nov 1 and never heard anything back, i tried leaving a voicemail for cory earlier today but have not gotten a call back yet, i hate not being able to deal with polycom directly |
| 22:07.50 | *** part/#asterisk BSDTech (n=RNeese@adsl-69-230-170-5.dsl.irvnca.pacbell.net) |
| 22:07.55 | *** join/#asterisk BSDTech (n=RNeese@adsl-69-230-170-5.dsl.irvnca.pacbell.net) |
| 22:08.03 | BSDTech | when is 1.4 now due out |
| 22:08.16 | costalivan | I'm using ekiga and I'm trying to register in ekiga.net |
| 22:08.20 | BSDTech | I thought it would be end of dec |
| 22:08.24 | costalivan | this is my network diagram |
| 22:08.25 | BSDTech | nov sorry |
| 22:08.46 | BSDTech | but I see its only beta 3 not even a rc |
| 22:09.13 | costalivan | 192.168.1.200 --> Linksys NAT to --- > 10.6.3.192 --->10.6.250.21 siproxd |
| 22:09.23 | costalivan | but it doesn't work |
| 22:09.32 | costalivan | the request arrive the 10.6.250.21 ip |
| 22:09.39 | costalivan | but after that nothing happens |
| 22:10.23 | BSDTech | nat: If you are behind a NAT you probably need to create an /etc/asterisk/sip_nat.conf file with AT LEAST these two lines: 1) externip=your.external.dotted.IPaddess 2) localnet=192.168.0.0/255.255.255.0 (assuming your local network uses 192.168.0.x addresses). Then sip reload from the CLI. |
| 22:10.33 | BSDTech | in this case in sip.conf |
| 22:10.54 | BSDTech | -or add a #include => sip_nat.conf in sip.conf |
| 22:13.41 | *** part/#asterisk costalivan (n=ivan@189.145.28.195) |
| 22:13.50 | bluregard | [TK]D-Fender: you got a minute? |
| 22:14.37 | *** join/#asterisk mega (n=mega@217.201.136.14) |
| 22:14.41 | [TK]D-Fender | bluregard : Shoot |
| 22:15.16 | *** join/#asterisk Winkie (n=urmom@cpc2-stre3-0-0-cust344.bagu.cable.ntl.com) |
| 22:15.58 | *** join/#asterisk le_nec (i=lldf@214.Red-81-37-87.dynamicIP.rima-tde.net) |
| 22:16.05 | le_nec | hi all |
| 22:16.17 | le_nec | have a problem |
| 22:16.21 | le_nec | can help me |
| 22:17.11 | bluregard | [TK]D-Fender: I've got the 501 set up with the first 2 line keys as my extension with the third monitoring my "0" ext. If I'm on a call on my ext. and someone calls my ext, it goes right to n+101 which is my busy VM. However, if someone calls the "0" ext it rings through just fine. |
| 22:17.18 | Math` | le_nec:with so little information, no |
| 22:17.21 | le_nec | i'm installed asterisk 1.4 and i can't recive call, when i recived a call, this is the error handle_request_invite: Failed to authenticate user |
| 22:17.42 | bluregard | [TK]D-Fender: any work around you know of? |
| 22:17.42 | le_nec | chan_sip.c:7916 check_auth: username mismatch, have <trunk_1>, digest has <s> |
| 22:18.14 | [TK]D-Fender | bluregard : redescribe the reg's, and line key usage. |
| 22:18.44 | bluregard | [TK]D-Fender: you want me to pastebin my phone1.cfg? |
| 22:18.53 | bluregard | it's pretty short |
| 22:18.55 | [TK]D-Fender | bluregard : Sure |
| 22:19.13 | [TK]D-Fender | bluregard : Change only the passwords |
| 22:19.39 | bluregard | no passwords in it |
| 22:21.00 | le_nec | anybody can help me, please? |
| 22:22.12 | bluregard | [TK]D-Fender: http://pastebin.ca/259974 |
| 22:22.57 | [TK]D-Fender | bluregard : EW |
| 22:23.19 | bluregard | huh |
| 22:23.22 | [TK]D-Fender | bluregard : Did that via the web interface, or direct on the phone? |
| 22:23.51 | bluregard | [TK]D-Fender: mostly the phone, a few things in the file itself |
| 22:24.18 | bluregard | [TK]D-Fender: I just noticed a mistake in it too |
| 22:24.21 | *** join/#asterisk mega_ (n=mega@217.201.133.147) |
| 22:24.33 | [TK]D-Fender | bluregard : Several... |
| 22:24.50 | [TK]D-Fender | bluregard : You really should do this phone over top-to-bottom |
| 22:25.18 | bluregard | [TK]D-Fender: I have all the original configs backed up |
| 22:25.47 | [TK]D-Fender | bluregard : You should only be using reg's 1 & 2, using 2 line-keys on the first, 1 on the second. |
| 22:26.11 | bluregard | [TK]D-Fender, That's what I tried first, this was the only way I could get it to work with more than just my ext on all 3 line keys |
| 22:26.17 | [TK]D-Fender | bluregard : and while you're at it maybe increase the # calls per line key allowed. Technically I'd just use 1 line key each allowing 5 calls per. |
| 22:26.43 | *** part/#asterisk jay (n=jay@lindalane.com) |
| 22:30.52 | brian | http://rafb.net/paste/results/CPEVXQ82.html |
| 22:31.08 | brian | Is there anything wrong with that that might screw up? |
| 22:32.43 | *** join/#asterisk [hC] (n=hardcore@66.119.169.94) |
| 22:34.39 | [TK]D-Fender | brian : exten => 8,1,Goto(_X.|3) <- not legal. it will attempt a pattern match on your GOTO and fail. you have to actually goto a number that would be matched by where you're looking to go, and even using a catch all like that is seriously ugly at best. |
| 22:34.52 | brian | it works |
| 22:35.13 | brian | i think |
| 22:35.26 | *** join/#asterisk jm|home (n=jamiem@dilbert.jamiem.com) |
| 22:36.11 | *** part/#asterisk le_nec (i=lldf@214.Red-81-37-87.dynamicIP.rima-tde.net) |
| 22:37.39 | *** join/#asterisk KeNroM (n=sdfgas@63.175.158.33) |
| 22:38.09 | brian | [TK]D-Fender: How do I do it correctly? |
| 22:38.43 | *** join/#asterisk ontae (n=root@clnet-p03-090.ikbnet.co.at) |
| 22:40.02 | KeNroM | where can i get some vicidail help? |
| 22:40.32 | *** join/#asterisk tr2x (n=alvar@80-218-150-90.dclient.hispeed.ch) |
| 22:41.30 | bluregard | damn, I can't believe I did that |
| 22:44.35 | *** join/#asterisk [hC] (n=hardcore@66.119.169.79) |
| 22:45.06 | brian | http://rafb.net/paste/results/CPEVXQ82.html <--- How do I fix this so I do it "correctly" |
| 22:46.55 | bluregard | [TK]D-Fender: http://pastebin.ca/259998 better? |
| 22:47.09 | *** join/#asterisk backblue (n=moo@87-196-68-22.net.novis.pt) |
| 22:47.11 | *** join/#asterisk sbingner (n=thanotos@pdpc/supporter/sustaining/sbingner) |
| 22:47.39 | sbingner | so I have an IAXY that keeps having to have the power killed and plugged back in -- any ideas as to why that would be happening? |
| 22:47.55 | [TK]D-Fender | bluregard : Your configs warrant a complete rebuild undoing what was keyed directly on the phone. |
| 22:48.07 | [TK]D-Fender | bluregard : and doesn't address the # of calls per line-key |
| 22:48.28 | bluregard | [TK]D-Fender: reset to default on the phone? |
| 22:48.31 | [TK]D-Fender | bluregard : though it does help the problem of calls not spanning on yuour 1st reg like before. |
| 22:48.47 | [TK]D-Fender | bluregard : Yes, and rebuild your sip.cfg and phoneXXXXX.cfg |
| 22:49.29 | *** join/#asterisk mpruett (n=mpruett@24-240-203-82.static.stls.mo.charter.com) |
| 22:49.43 | bluregard | [TK]D-Fender: my sip.cfg shouldn't have changed other than what I put in with the text editor though should it? |
| 22:51.27 | *** part/#asterisk andresmujica (n=andresmu@201.244.199.222) |
| 22:51.28 | *** join/#asterisk jm|home (n=jamiem@dilbert.jamiem.com) |
| 22:51.58 | lilalinux | I have an old NTBA here where I don't know which pin is a1,a2,b1,b2, how can I find out? |
| 22:52.10 | lilalinux | There is one pin numbered 1 |
| 22:52.16 | bluregard | [TK]D-Fender: reset local config or device setting? |
| 22:52.20 | [TK]D-Fender | bluregard : I'm just betting based on how much you have in overriders that you didn't populate any of the files in the way you really should have. |
| 22:52.21 | lilalinux | is there a standard? |
| 22:52.40 | [TK]D-Fender | bluregard : before you flush the phone, you should ahve a full set of rebuilt configs for it. |
| 22:53.41 | bluregard | [TK]D-Fender: All I changed in sip.cfg was for the idle bitmap, in <mac>.cfg |
| 22:53.49 | *** join/#asterisk mega_ (n=mega@217.201.140.159) |
| 22:55.40 | Shaun2222 | i keep getting a confg file error 0x4020 |
| 22:55.43 | Shaun2222 | anybody know what that means? |
| 22:56.57 | *** join/#asterisk Archi1999 (n=mlsmith@adsl-70-247-240-222.dsl.ltrkar.swbell.net) |
| 22:57.52 | Archi1999 | Can anybody here help me with snap? |
| 22:57.59 | [TK]D-Fender | bluregard : sip.cfg should have your server, SNTP, dialplan, and a pile of other general settings. phoneXXX.cfg should have your username, pass, and line key layout, nothing more. <mac>-phone.cfg should be EMPTY |
| 22:58.03 | Archi1999 | snapanumber |
| 22:58.11 | *** part/#asterisk danbrwn (n=danny@216.77.58.40) |
| 22:59.00 | bluregard | [TK]D-Fender: ok |
| 22:59.30 | *** join/#asterisk alex_112 (n=admin@fw.packetfront.com) |
| 22:59.37 | Aurs | Shaun2222: http://lists.digium.com/pipermail/asterisk-users/2006-April/147968.html |
| 23:02.14 | Aurs | ..did not help much |
| 23:02.21 | Shaun2222 | nopr |
| 23:02.38 | Shaun2222 | whats weird is i have another phone using this exact config |
| 23:02.40 | Shaun2222 | working fine |
| 23:02.43 | *** join/#asterisk nyt (i=nyt@countercultured.net) |
| 23:03.04 | Shaun2222 | the phone also never downloads the phone.cfg or sip.cfg |
| 23:03.19 | Shaun2222 | i'm telling it to use ftps, wonder if it's some type of problem with that |
| 23:03.42 | Aurs | on the same ftp? |
| 23:03.56 | Shaun2222 | the other one is using ftp... |
| 23:04.00 | *** join/#asterisk atapi (n=virgill4@c-65-34-182-167.hsd1.fl.comcast.net) |
| 23:04.01 | Shaun2222 | i decided to try ftps on this one |
| 23:04.14 | *** join/#asterisk henrique (n=henrique@201-26-77-205.dsl.telesp.net.br) |
| 23:04.34 | Aurs | never tried ftps |
| 23:04.52 | Aurs | or https |
| 23:05.08 | *** part/#asterisk atapi (n=virgill4@c-65-34-182-167.hsd1.fl.comcast.net) |
| 23:05.44 | *** join/#asterisk ontae (n=ontae@clnet-p03-090.ikbnet.co.at) |
| 23:05.59 | Shaun2222 | well |
| 23:06.09 | Shaun2222 | it's downloading the sip.cgi and phone.cfg |
| 23:06.12 | Shaun2222 | guess sftp doesnt work. |
| 23:06.22 | Shaun2222 | so much for being secure. |
| 23:06.38 | Aurs | by sip.cgi you mean sip.cfg I guess |
| 23:06.46 | Shaun2222 | ya, typo :) |
| 23:06.50 | *** join/#asterisk stuq (n=stuq@user-12lcqia.cable.mindspring.com) |
| 23:07.03 | Aurs | does it try to fetch 00000000.cfg? |
| 23:07.08 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.mn.comcast.net) |
| 23:07.16 | Aurs | (000000000000.cfg) |
| 23:07.29 | Shaun2222 | no, it went straight for mac.cfg |
| 23:07.48 | ontae | [5~[5~[6~qquit |
| 23:07.57 | Shaun2222 | mac.cfg tells it to load sip.ld.ver and then mac-phone.cfg and mac-sip.cfg |
| 23:08.15 | *** join/#asterisk vader-- (n=johndoe@204.183.88.101) |
| 23:08.43 | Aurs | could be that the encryption screws things up, I guess |
| 23:08.44 | sbingner | anybody know of a way to force an iaxy to reload its firmware? |
| 23:08.45 | vader-- | hola |
| 23:08.54 | Aurs | but that sounds strange too on second thought |
| 23:09.15 | Aurs | because it reads the mac.cfg correct, if it fetches the files in CONFIG-FILES="" in that one... |
| 23:09.36 | vader-- | anyone recommend a solution for this situation. We have been having issues where a person will go to dial extension 11x, i.e. 114 and accidently dial 9 then 114 and the system reads that as 911 because as soon as it sees the 911 it dials |
| 23:09.48 | Shaun2222 | well with ftps, it fetched mac.cfg and stored mac-boot.log |
| 23:09.49 | vader-- | what are the legalities of messing with the way 911 is handled |
| 23:09.50 | Shaun2222 | then crashed |
| 23:10.04 | *** join/#asterisk ontae (n=ontae@clnet-p03-090.ikbnet.co.at) |
| 23:10.32 | vader-- | here are a couple ideas i had, make it so 9911 is the only way to get to 911, make it ask a message like, are you sure you want to dial 911, or have it play a message when 911 is dialed that says you must first dial a 9 |
| 23:10.54 | Aurs | vader--: do you have to confirm dialing 911 on other phones? sounds like a bad idea |
| 23:12.29 | Aurs | Shaun2222: ok, so it did NOT fetch mac-sip.cfg when using ftps? that could mean that it could not read the mac.cfg correctly |
| 23:12.58 | Shaun2222 | Aurs: it looks like it downloads mac.cfg and has a issue reading it |
| 23:13.10 | Aurs | perhaps you need to tweak your sshd_conf in some way? |
| 23:13.12 | Shaun2222 | works fine with ftp just not ftps |
| 23:13.25 | backblue | are you speaking about polycoms? |
| 23:13.31 | Aurs | backblue: yes |
| 23:13.43 | backblue | but it does not use sftp, dont you mean tftp? |
| 23:13.49 | Shaun2222 | ftps is not ssh...it's still through the ftp server... your thinking of ftp over ssh2 |
| 23:14.08 | backblue | yes it's not sftp over ssh |
| 23:14.11 | Shaun2222 | backblue: it gives ftp/ftps/http/https/tftp as options. |
| 23:14.17 | backblue | yes |
| 23:14.18 | Aurs | ah, ok |
| 23:14.25 | Aurs | i thought it was ssh |
| 23:14.29 | backblue | but you were speaking about sftp |
| 23:14.39 | backblue | ftps != sftp |
| 23:14.56 | Aurs | agreed |
| 23:15.23 | Shaun2222 | ya ya, one type over the other 10 times i said ftps... |
| 23:15.24 | Shaun2222 | :) |
| 23:15.29 | Aurs | sftp > ftps |
| 23:15.30 | Aurs | :P |
| 23:16.05 | Aurs | ftps is SSL+ftp, right? |
| 23:16.29 | Shaun2222 | right. |
| 23:16.32 | Aurs | wouldn't surprise me if you need a polycom-signed certificate or something ;) |
| 23:17.07 | Shaun2222 | polycom does have some ssl options for ca certs and whatnot |
| 23:17.10 | Shaun2222 | could be |
| 23:17.11 | Aurs | or at least _a_ signed cert |
| 23:17.28 | Shaun2222 | honesly i've never seen a ftp_server or client complain about a selfsigned cert.. |
| 23:17.32 | Shaun2222 | but could happen. |
| 23:17.59 | Aurs | it's security.. it's not supposed to be simple |
| 23:18.01 | Aurs | hehe |
| 23:18.02 | Shaun2222 | either way you think it would give me a better error |
| 23:18.23 | Aurs | if you could find 0x4something in the polycom docs, it would be ok |
| 23:18.37 | Aurs | but I guess you have grep'd the docs ;) |
| 23:21.08 | Shaun2222 | hmm, ok to the next issue for now.. |
| 23:21.24 | Shaun2222 | i want to have multiple phones view the same extention (line).. |
| 23:21.38 | Shaun2222 | right now i have both phones configured for ext301 for testing... |
| 23:21.47 | Shaun2222 | if one phone picks up the line, the other phone doesnt see that. |
| 23:22.14 | Shaun2222 | anyway to make it show what lines are being used? |
| 23:22.27 | Aurs | 2.2.3 in the admin guide |
| 23:22.38 | Aurs | 2.2.3 Management of File Encryption and Decryption |
| 23:22.57 | Aurs | page 23 |
| 23:23.19 | Aurs | The device.sec.configEncryption.key configuration file |
| 23:23.19 | Aurs | parameter is used to set the key on the phone. |
| 23:23.49 | Shaun2222 | hmm both phones dont ring either when dialing the same extention... |
| 23:23.59 | Shaun2222 | looks like one phone is taking over the other |
| 23:24.42 | Aurs | and yes, it trusts certificates signed by authorities |
| 23:25.01 | *** join/#asterisk underzsof (n=cvdsfg@ppp158-144.adsl.forthnet.gr) |
| 23:25.04 | underzsof | THE SITE HAS EVERYTHING ABOUT WAREZ RAPIDSHARE DOWNLOADZ --> WWW.UNDERZSOFT.COM THANX!!! |
| 23:25.07 | Aurs | so buy yourself a signed SSL cert and it will work |
| 23:25.12 | sbingner | lol |
| 23:25.20 | underzsof | THE SITE HAS EVERYTHING ABOUT WAREZ RAPIDSHARE DOWNLOADZ --> WWW.UNDERZSOFT.COM THANX!!! |
| 23:25.22 | *** part/#asterisk underzsof (n=cvdsfg@ppp158-144.adsl.forthnet.gr) |
| 23:25.29 | sbingner | Aurs: it has no mechanism to load another CA cert? |
| 23:25.47 | hmmhesays | wow asterisk is freaking out |
| 23:25.57 | Aurs | sbingner: yes, it does. page 74 in the admin guide pdf |
| 23:25.59 | hmmhesays | 137 active calls show on the console |
| 23:26.06 | hmmhesays | when there are only 2 |
| 23:26.11 | hmmhesays | 2 actual calls |
| 23:26.14 | Aurs | hmmhesays: queue calls? |
| 23:26.58 | sbingner | hmmhesays: show channels to pastebin.ca? |
| 23:27.27 | Aurs | "In addition, custom certificates can be added to the phone. This is done by using the SSL Security menu on the phone to provide the URL of the custom certificate then select an option to use this custom certificate" |
| 23:27.48 | hmmhesays | no queues |
| 23:28.02 | Shaun2222 | is it not possible to have 2 phones registered to the same ext? |
| 23:28.10 | sbingner | Shaun2222: no, it's not |
| 23:28.25 | Shaun2222 | how would you configure say line1/2/3 on all phones |
| 23:28.28 | hmmhesays | show channels just shows a large amount of up calls |
| 23:28.30 | sbingner | Shaun2222: they don't register to extensions, they register as devices |
| 23:28.38 | Aurs | Shaun2222: but you can have ext0 and ext1 with the samme callerid in sip.conf |
| 23:28.42 | sbingner | Shaun2222: you can have an extention call multiple devices |
| 23:29.19 | Aurs | and if someone calls in to that ext, dial SIP/ext0&ext1 |
| 23:29.29 | Aurs | SIP/ext0&SIP/ext1 |
| 23:30.07 | sbingner | sip.conf = devices and extensions.conf = extensions |
| 23:30.33 | *** join/#asterisk jm|home (n=jamiem@dilbert.jamiem.com) |
| 23:32.24 | hmmhesays | i wonder wtf is causing this |
| 23:32.34 | Shaun2222 | ok, i basically want line1/2/3 on all phones, i also want it to show if a line is in use on each phone... |
| 23:32.46 | sbingner | Shaun2222: that's shared line presentation |
| 23:32.57 | sbingner | Shaun2222: I think it's beta in * currently |
| 23:33.06 | Shaun2222 | ok |
| 23:33.57 | sbingner | Shaun2222: see sla.conf in 1.4 apparently |
| 23:33.59 | sbingner | I haven't used it |
| 23:34.20 | [TK]D-Fender | Shaun2222 : What model of phone? |
| 23:34.33 | Shaun2222 | 601 |
| 23:35.48 | [TK]D-Fender | Shaun2222 : You want to check the status of actual phone LINES on *? |
| 23:36.23 | Shaun2222 | [TK]D-Fender: if one phone is using line1 i want the other phones to see line1 is in use.. |
| 23:37.24 | [TK]D-Fender | Shaun2222 : then add them as a watched buddy and set your hint to point to the channel. |
| 23:37.27 | Aurs | ok, on the buddy list? |
| 23:37.27 | sbingner | that's SLA, but I don't know how well * supports it currently |
| 23:37.50 | Aurs | Shaun2222: do you have the expansion module for the 601? |
| 23:38.00 | sbingner | and you want to hit "Line1" to pick up the call if somebody else put it on hold right? |
| 23:39.35 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 23:39.44 | [TK]D-Fender | Aurs : Yes, the buddy list... |
| 23:40.00 | sbingner | Shaun2222: in the 601 specs it references Shared call appearance, bridged line appearance (Key |
| 23:40.00 | sbingner | System emulation) |
| 23:40.03 | Shaun2222 | Aurs: no i dont have the addon module |
| 23:40.14 | sbingner | look for that in your polycom manual for how to set it up if you can |
| 23:40.14 | *** join/#asterisk DavoFrom818 (n=Vito310@69.163.90.220) |
| 23:40.15 | DavoFrom818 | hi |
| 23:40.18 | DavoFrom818 | how is everyone |
| 23:40.20 | Shaun2222 | sbingner: right. |
| 23:40.36 | Aurs | [TK]D-Fender: have you tried the expansion module on 601? |
| 23:40.43 | DavoFrom818 | am i in the right channel for support on AsteriskNOW? |
| 23:40.57 | Shaun2222 | i would really like to be able to see calls and where they are parked too from the phone. |
| 23:41.17 | Shaun2222 | the phone has a park button but i coudlnt get it to work, i only messed with it for a few minutes last night |
| 23:41.39 | *** join/#asterisk DavoFrom818 (n=Vito310@69.163.90.220) |
| 23:41.45 | DavoFrom818 | sorry i got dc |
| 23:42.02 | sbingner | Shaun2222: try setting up the devices in sla.conf and see if it works |
| 23:42.04 | DavoFrom818 | so is this the channel for *now? |
| 23:42.28 | sbingner | DavoFrom818: no, it's not but if it's not a question specific to *now we may be able to answe |
| 23:42.42 | [TK]D-Fender | Aurs : Yes I have. My receptionist has 2 of the,m |
| 23:42.48 | Shaun2222 | sbingner: i dotn have a sla.conf but i suppose i'll have to loko that one up. |
| 23:42.58 | sbingner | Shaun2222: are you running * 1.4? |
| 23:43.07 | lilalinux | when I want to use zaphfc, do I still need the hisax drivers? |
| 23:43.08 | sbingner | Shaun2222: it's in the sample configs for 1.4 |
| 23:43.10 | Shaun2222 | 1.2.13 |
| 23:43.17 | sbingner | aah for SLA you need 1.4 |
| 23:43.18 | [TK]D-Fender | sbingner : SLC, and SCA won't do you much good when you want to monitor what is most likely a Zap device. |
| 23:43.27 | Shaun2222 | 1.4 is beta isnt it? |
| 23:43.37 | sbingner | [TK]D-Fender: he said he's using polycom |
| 23:43.39 | [TK]D-Fender | sbingner : Doesnt' work that way. SLA, and SCA are for watching PHONES. |
| 23:43.56 | *** join/#asterisk DavoFrom818 (n=Vito310@69.163.90.220) |
| 23:44.00 | sbingner | not SIP polycom? |
| 23:44.03 | DavoFrom818 | omg what is happening to my connection |
| 23:44.08 | [TK]D-Fender | sbingner : I know full well what he's using, and like I said those 2 features weren't designed in* to work like that. |
| 23:44.13 | [TK]D-Fender | *sigh* |
| 23:44.36 | DavoFrom818 | any ideas? |
| 23:44.51 | sbingner | [TK]D-Fender: the only way he will get what he wants is SLA, if it doesn't work yet then you must be right |
| 23:45.54 | Aurs | [TK]D-Fender: and you can have how many "watched buddies" on one module? |
| 23:45.57 | sbingner | DavoFrom818: no, it's not but if it's not a question specific to *now we may be able to answer -- we never got a question |
| 23:46.06 | [TK]D-Fender | sbingner : Again, its not made to let you grab LINES, its meant for you to grab PHONES. |
| 23:46.20 | sbingner | [TK]D-Fender: right... your point? |
| 23:46.38 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
| 23:46.39 | [TK]D-Fender | Aurs : You can watch all of them. Which at 14 per * 3 = 42 +6 on base = 48 max |
| 23:46.46 | DavoFrom818 | it must of cut me off |
| 23:46.53 | sbingner | the polycom doesnt register as multiple devices to do multiple lines? |
| 23:47.00 | DavoFrom818 | ok let me reset my switch before i continue brb sbingner thnx |
| 23:47.08 | [TK]D-Fender | sbingner : Point is that there is no way to use SLA/SCA to monitor line status and grab a line direct. |
| 23:47.29 | *** join/#asterisk robin_sz (n=robin@rapid2.gotadsl.co.uk) |
| 23:48.00 | sbingner | [TK]D-Fender: but it would let you pick it up when it was placed on hold, no? if not then SLA isn't fully supported yet, which was my original disclaimer ;) |
| 23:48.08 | [TK]D-Fender | sbingner : because its not a direct channel. you go from a SIP channel to a ZAP by letting * pick the resource to use. if you want to monitor a line, then Presence is what you're looking for, but there is no way to "pull" a line" specifically with some seriously ugly hacks |
| 23:48.29 | [TK]D-Fender | sbingner : You are just not following what kind of devices it was inteded to be used with. |
| 23:48.34 | Aurs | [TK]D-Fender: but sometimes the status on buddies is "stalling", and I have to reboot the phone, or disable/enable watch buddy to get it to work again. guess I would choose reboot if I had 48 buddies ;) |
| 23:48.56 | [TK]D-Fender | sbingner : You can watch someone elses phone and grab a call they put on hold, but don't expect to pull that off with a Zap line. |
| 23:49.24 | sbingner | [TK]D-Fender: the example shows both zap and sip |
| 23:49.42 | [TK]D-Fender | Aurs : only reason it would stall is if you're on an old release (pre 1.2.5 or so) where "reload" killed Presencesubscriptions. |
| 23:49.43 | sbingner | [TK]D-Fender: anyways I'll have to play with it sometime heh |
| 23:49.47 | *** join/#asterisk dr0ne (n=fn@CABLE-72-53-45-212.cia.com) |
| 23:49.51 | *** join/#asterisk ltd (n=z@202-161-1-61.dyn.iinet.net.au) |
| 23:50.38 | Aurs | [TK]D-Fender: ok, perhaps that has improved (running 1.2.13 now). unless restarting asterisk stalls presence on polycoms |
| 23:50.50 | [TK]D-Fender | Aurs : Not that I've seen. |
| 23:51.04 | [TK]D-Fender | Aurs : And I am running 30 of them on 1.2.13 |
| 23:51.37 | Aurs | [TK]D-Fender: ok. been a while since I had to "rewatch" or reboot, when I think about it |
| 23:52.02 | *** join/#asterisk DavoFrom818 (n=Vito310@69.163.90.220) |
| 23:52.07 | DavoFrom818 | sbingner ok i hope that worked |
| 23:52.35 | DavoFrom818 | sbingner the problem i am having is when i go to add a service provider the provider box is empty |
| 23:52.40 | *** join/#asterisk malph (n=malph@66-231-0-194.hosts.sdnet.net) |
| 23:52.55 | DavoFrom818 | sbingner any idea? |
| 23:53.04 | sbingner | DavoFrom818: yea you'll need to find the actual *now support for that, it's specific to their web interface |
| 23:53.23 | DavoFrom818 | sbingner what room are they in? |
| 23:53.42 | sbingner | I have no idea, they may not have an IRC support channel |
| 23:53.43 | [hC] | is it possible to do what immediate=yes does in zapata.conf, but for a sip phone? |
| 23:54.03 | sbingner | [hC]: it depends on the phone, some do support that but it's on the phone |
| 23:54.05 | DavoFrom818 | sbingner what is the best gui configure for asterisk? |
| 23:54.26 | sbingner | DavoFrom818: sorry I can't answer that, I don't use any of the GUIs |
| 23:54.32 | malph | I'm receiving an error on the cli and I am having trouble finding an answer with google can I post it here in hopes that someone recognizes it? |
| 23:55.20 | DavoFrom818 | sbingner where can i get a howto on doing it manually? |
| 23:55.38 | sbingner | DavoFrom818: examples are included with the asterisk installation |
| 23:56.02 | sbingner | DavoFrom818: and http://www.voip-info.org/wiki-Asterisk has alot of goof indo |
| 23:56.39 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 23:57.19 | [TK]D-Fender | [hC] : Depends on the phone. |
| 23:57.41 | lilalinux | doesn't the debian package asterisk-bristuff bring the zaphfc module? I can't find it anywhere |
| 23:58.27 | bluregard | [TK]D-Fender: which reset should I do? local config, device settings or both? |
| 23:59.11 | sbingner | [hC]: see Private Line Automated Ringdown (PLAR) |