| 00:00.08 | SpeedEvil | http://www.ebay.co.uk/itm/New-56K-USB-V9-0-V-92-External-Dial-Up-Voice-Fax-Data-Modem-for-Win7-Windows7-64-/261278877198?pt=UK_Computing_Networking_SM&hash=item3cd56f460e |
| 00:00.13 | SpeedEvil | Claims to support linux. |
| 00:00.20 | SpeedEvil | Why do I suspect this may not be simple. :) |
| 00:00.32 | WIMPy | Maybe, but certainly not Asterisk. |
| 00:00.43 | WIMPy | Asterisk can't use modems. |
| 00:01.17 | SpeedEvil | Seems like in principle it'd just be a simple driver. |
| 00:01.36 | SpeedEvil | (assuming for a moment there is a defacto voice standard) |
| 00:01.41 | navaismo | WIMPy, only old modems with tiger chipsets asterisk see it as x100p |
| 00:02.01 | WIMPy | For an USB modem? Extremely unlikely. |
| 00:02.17 | navaismo | ah sorry i was talking about internal |
| 00:02.18 | navaismo | my bad |
| 00:02.31 | WIMPy | I know DAHDI sees them, but does that actually work? |
| 00:02.57 | navaismo | SpeedEvil, if you know how to develop a driver for linux, patch the dahdi and asterisk then yes is "only matter of drivers" |
| 00:03.02 | navaismo | WIMPy, yes, |
| 00:03.14 | navaismo | but you will have echo |
| 00:03.30 | navaismo | SpeedEvil, your best option is sangima u100 |
| 00:05.25 | SpeedEvil | It's depressing that for much USB stuff it seems the better option is almost to say screw-it and build your own device. |
| 00:05.34 | SpeedEvil | (hitting related issues around audio) |
| 00:06.46 | navaismo | LMAOing |
| 00:07.19 | navaismo | when you develop such solution you will see the meaning of "expensive" |
| 00:07.23 | SpeedEvil | Approvals would be unfortunately annoying. |
| 00:07.28 | SpeedEvil | The actual hardware is not. |
| 00:07.59 | navaismo | but take here go ahead finish and share with all of us-->http://openusbfxo.wordpress.com/ |
| 00:13.26 | SpeedEvil | And yes, I'm quite aware of the issues around boards. |
| 00:13.34 | SpeedEvil | http://www.mauve.plus.com/opensourcehw.txt |
| 00:19.03 | *** join/#asterisk saint_ (~saint@c-50-166-85-78.hsd1.nj.comcast.net) |
| 00:30.53 | *** join/#asterisk Milarepa (~Milarepa@host-74-211-92-125.beyondbb.com) |
| 00:32.00 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 00:35.09 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 00:36.48 | *** part/#asterisk bulkorok (~Benjamin@gw1.pinguin.ag) |
| 00:49.35 | *** join/#asterisk calum_ (~calum_@cpc4-harg5-2-0-cust371.7-1.cable.virginm.net) |
| 01:01.39 | WIMPy | Y |
| 01:03.43 | *** join/#asterisk newtonr (~newtonr@173-17-135-67.client.mchsi.com) |
| 01:03.44 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 01:22.24 | *** join/#asterisk eoss (~chatzilla@pdpc/supporter/student/eoss) |
| 01:22.41 | eoss | How would one verify if port 5060 is open if we are using UDP for SIP |
| 01:23.44 | [TK]D-Fender | Depends who "one" is |
| 01:24.39 | *** join/#asterisk k-man (~jason@unaffiliated/k-man) |
| 01:44.01 | *** join/#asterisk brendan` (~textual@107-0-240-226-ip-static.hfc.comcastbusiness.net) |
| 01:50.14 | *** join/#asterisk jasonwert (~w3rt@71.89.137.28) |
| 01:50.48 | *** join/#asterisk tonyclewis (uid6025@gateway/web/irccloud.com/x-wqparlhsuuzbsxjl) |
| 01:53.38 | eoss | [TK]D-Fender: one is capable of learning, so any suggestions would be much appreciated |
| 01:56.52 | [TK]D-Fender | <PROTECTED> |
| 01:57.12 | [TK]D-Fender | if it's YOUR server you can tell if it's "open" |
| 01:57.27 | [TK]D-Fender | Then it's another matter if your server RESPONDS to the request |
| 02:09.29 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 02:11.02 | *** join/#asterisk darkdrgn2k (~darkdrgn2@69-165-131-20.dsl.teksavvy.com) |
| 02:11.06 | darkdrgn2k | evening all |
| 02:11.41 | *** join/#asterisk jploh (~textual@49.146.167.241) |
| 02:17.43 | eoss | [TK]D-Fender: It's not a server I have access to, but it should be accepting SIP requests over UDP |
| 02:18.25 | eoss | I just want to ensure that a particular location can reach and utilize this server on port 5060 over UDP |
| 02:18.32 | [TK]D-Fender | eoss: You need access. UDP has no "psychic mode". If the server isn't responding, perhaps it's not even getting the request... or it is failing to route a response. |
| 02:21.01 | eoss | Yes, that's a possibility, but we know that this server responds and works in other locations, the point is to prove that it's NOT working at this location by somehow simulating a sip udp connection and watching it fail for one reason or the other, or work |
| 02:21.43 | eoss | I guess I will go 1337 h4x0r and craft my own UDP packet via python to see if I get a response |
| 02:22.09 | WIMPy | sipsak? |
| 02:23.47 | eoss | WIMPy: thanks that looks like it could work |
| 02:25.44 | *** join/#asterisk tuxd00d (~tuxd00d@ip24-251-34-116.ph.ph.cox.net) |
| 02:34.13 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 02:37.08 | *** join/#asterisk elguero (~miguel323@2001:470:1f06:12c4::2) |
| 02:59.51 | *** join/#asterisk micols (~gnu@shell1.rlogin.dk) |
| 03:06.25 | *** join/#asterisk koffel (koffel@173-167-212-107-ip-static.hfc.comcastbusiness.net) |
| 03:06.42 | *** join/#asterisk jploh (~textual@49.146.167.241) |
| 03:19.16 | koffel | anyone in here use polycom ip? |
| 03:25.30 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 03:26.49 | *** join/#asterisk infernixx (nix@unaffiliated/infernix) |
| 03:28.36 | WIMPy | ~polls |
| 03:28.36 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
| 03:36.40 | koffel | i have a polycom ip 320 and i get this error chan_sip.c:14399 check_auth: username mismatch, have <x>, digest has <x1>? |
| 03:37.07 | *** join/#asterisk D30 (~deo@203.177.9.66) |
| 03:44.44 | *** join/#asterisk brendan` (~textual@107-0-240-226-ip-static.hfc.comcastbusiness.net) |
| 03:51.36 | *** join/#asterisk jploh (~textual@49.146.167.241) |
| 04:27.11 | *** join/#asterisk D30 (~deo@203.177.9.66) |
| 04:35.01 | *** join/#asterisk vinhdizzo (~vinh@cpe-98-154-220-20.socal.res.rr.com) |
| 05:18.06 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
| 05:18.28 | *** join/#asterisk tonyclewis (uid6025@gateway/web/irccloud.com/x-qlbnoytdmkbzhyut) |
| 05:35.13 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 05:35.15 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 06:03.20 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 06:09.53 | *** join/#asterisk CeBe (~CeBe@port-92-206-84-112.dynamic.qsc.de) |
| 06:14.20 | *** join/#asterisk bkruse (~Adium@24.42.229.8) |
| 06:21.34 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 06:31.06 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 06:39.37 | *** join/#asterisk D30 (~deo@203.177.9.116) |
| 06:42.14 | *** join/#asterisk D30_ (~deo@203.177.9.66) |
| 06:49.44 | *** join/#asterisk D30 (~deo@58.71.19.181) |
| 06:50.15 | *** join/#asterisk D30_ (~deo@58.71.19.178) |
| 07:13.34 | *** join/#asterisk bkruse (~Adium@24.42.229.8) |
| 07:21.56 | *** join/#asterisk D30_ (~deo@58.71.19.181) |
| 07:23.34 | *** join/#asterisk D30__ (~deo@222.127.13.226) |
| 07:35.37 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 07:54.05 | *** join/#asterisk jhlavacek (~jirka@78.208.220.3) |
| 07:54.06 | *** join/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 07:54.43 | bulkorok | hi |
| 08:06.51 | *** join/#asterisk Blashyrkh (~michaeld@94.136.25.97) |
| 08:07.23 | Blashyrkh | i have a problem with asterisk 11.8 and iaxmoden. Its not connecting. Settings and everything are doublechecked |
| 08:08.01 | Blashyrkh | when i start iaxmode i get a register timout, and with netstat see the recvq from the asterisk process getting higher |
| 08:12.39 | *** join/#asterisk cmendes0101 (~cmendes01@pool-96-251-59-96.lsanca.fios.verizon.net) |
| 08:26.16 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:d5c5:773c:d278:a69b) |
| 08:30.37 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 08:31.13 | Kobaz | so umm, shouldn't [authentication] deny=0.0.0.0/0.0.0.0 permit=192.168.50.0/24 apply to all peers? |
| 08:40.56 | Kobaz | i guess it only works per-peer |
| 08:46.51 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 09:04.52 | *** join/#asterisk tedstriker (~tedstrike@host-135-196-33-208.lines.viateldsl.com) |
| 09:07.46 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 09:37.01 | *** join/#asterisk oatha (~oatha@unaffiliated/athayde) |
| 09:38.24 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 09:40.31 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 09:40.41 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 09:42.08 | *** join/#asterisk vlad_sta_ (~vlad_star@195.218.175.34) |
| 09:50.58 | *** join/#asterisk boratynskikamil (~kamilbora@wpb89.bialnet.pl) |
| 09:51.03 | boratynskikamil | Good morning. |
| 09:56.08 | boratynskikamil | Is it possible to change monitor file format in Queue? |
| 09:59.49 | *** join/#asterisk Takapa (vegard@junior.svanberg.no) |
| 10:02.27 | boratynskikamil | I mean, filename. |
| 10:02.40 | boratynskikamil | Not file type |
| 10:09.07 | *** join/#asterisk kaotiko (~0xFF@unaffiliated/kaotiko) |
| 10:10.14 | *** join/#asterisk wonderworld (~ww@ip-62-143-158-113.unitymediagroup.de) |
| 10:13.14 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 10:21.07 | *** join/#asterisk VSpike (~johncc@37.48.225.145) |
| 10:24.33 | VSpike | Hi. I have system set up and running AsteriskNOW 2.0.0. It hasn't had any updates applied for some time, either OS or modules. I'm wondering, without a re-install what version can it update itself to? Will it go up through 2.x versions for example? How long could this version be considered "supported" in some way? |
| 10:25.20 | VSpike | I should probably do a clean install of 3.0 at some point but just weighing the options |
| 10:25.30 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 10:26.11 | VSpike | There's no VoIP so the box is not publicly visible - uses ISDN and an IAX trunk over VPN |
| 10:27.49 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
| 10:32.41 | *** join/#asterisk undecided (~kvirc@109.66.0.140) |
| 10:35.54 | undecided | just mograted from asterisk 8.* to 12 |
| 10:36.00 | undecided | migrated |
| 10:36.23 | undecided | problem |
| 10:37.49 | undecided | whenever I type asterisk commands on BASH terminal (asterisk -rx "dialplan reload") output says "Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 10:37.50 | undecided | " |
| 10:39.08 | undecided | asterisk.ctl doesn't exist on my system, but asterisk is running |
| 10:39.21 | undecided | what can be done? |
| 10:41.28 | mirela666 | undecided: might be that * is running as root and you are not |
| 10:41.54 | mirela666 | run ps aux | grep asterisk |
| 10:42.02 | mirela666 | run: |
| 10:42.21 | mirela666 | and whoami |
| 10:44.34 | undecided | I'm root |
| 10:44.50 | undecided | all run under root |
| 10:45.29 | undecided | why i don't have the /var/run/asterisk/asterisk.ctl ? |
| 10:47.07 | wdoekes | undecided: check asterisk.conf for the varrundir path |
| 10:47.30 | wdoekes | *astrundir |
| 10:48.35 | undecided | astrundir => /var/run/asterisk |
| 10:58.07 | boratynskikamil | Damn. If I added 0 same => Set(MONITOR_FILENAME=ISDN_)¬ |
| 10:58.10 | boratynskikamil | <PROTECTED> |
| 10:58.30 | boratynskikamil | such as line, shouldn't my recording look like: "ISDN_"? |
| 11:05.47 | *** join/#asterisk mattcen (~mattcen@c110-22-201-130.sunsh4.vic.optusnet.com.au) |
| 11:10.56 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 11:14.38 | wdoekes | undecided: (a) either your asterisk.conf is not read (is it looking in /usr/local/etc?), or (b) the [directories] are a template (does it have a trailing (!)), or (c) something prevents the socket from getting initialized |
| 11:15.23 | undecided | wdoekes: np it's working now |
| 11:15.36 | undecided | undecided: thank you |
| 11:16.01 | wdoekes | what turned out to be the problem then? |
| 11:16.15 | undecided | I think it had something to do with the controlling shell |
| 11:16.56 | undecided | I couldn't use the 'asterisk -vvvvvvvr' for example |
| 11:17.29 | wdoekes | next time, stop asterisk and use strace: strace -ebind,open asterisk -c 2>&1 >/dev/null | egrep 'asterisk.ctl|asterisk.conf' |
| 11:18.02 | undecided | but "asterisk -vvvvvvvvvvc" worked |
| 11:18.21 | wdoekes | asterisk -c is completely different from -r |
| 11:18.28 | wdoekes | -c starts an instance in the foreground |
| 11:18.39 | wdoekes | -r opens a remote-control to a started instance |
| 11:18.55 | wdoekes | so if you have no asterisk running, -r will indeed complain |
| 11:19.10 | wdoekes | that's the whole idea |
| 11:19.19 | undecided | yea but I had to do this in order to access asterisk cli |
| 11:19.38 | undecided | anyway |
| 11:20.09 | undecided | when I typed "service asterisk restart" everything started to work fine |
| 11:20.20 | undecided | hope this problem won't reoccure |
| 11:20.25 | undecided | and thank you :) |
| 11:22.28 | ectospasm | undecided: don't run asterisk -c unless you're troubleshooting a specific problem |
| 11:22.55 | ectospasm | run asterisk by itself with no options to start it as the a daemon, or better yet start it from your init system |
| 11:23.24 | ectospasm | once asterisk is running you can connect to it with asterisk -r |
| 11:23.33 | ectospasm | that is the proper way to do it. |
| 11:23.45 | undecided | sure it's on init |
| 11:23.49 | ectospasm | if you run it with -c and the controlling tty goes down, so does asterisk |
| 11:24.23 | undecided | but not sure why I had those troubles |
| 11:25.12 | undecided | I couldn't access logger without -c option |
| 11:25.41 | undecided | and running asterisk commands on bash terminal didn't work |
| 11:25.48 | ectospasm | you were doing something wrong then |
| 11:26.02 | undecided | it said ctl file was missing |
| 11:26.04 | undecided | no |
| 11:26.14 | undecided | it's a fresh install on vps |
| 11:26.23 | ectospasm | did you install Asterisk? |
| 11:26.32 | ectospasm | or was it pre-installed? |
| 11:26.58 | undecided | I built it from source |
| 11:27.59 | ectospasm | yeah, I haven't hit the missing control file problem in a while, I may not remember how to workaround it |
| 11:29.00 | undecided | yea Google is full of people complaining about it sometimes |
| 11:29.19 | boratynskikamil | <PROTECTED> |
| 11:29.24 | undecided | seems this problem has various reasons |
| 11:35.00 | undecided | boratynskikamil: that looks ok. Do ypu get any error? |
| 11:39.04 | boratynskikamil | undecided: Nope, I didn't reloaded dialplan, fail. :-) |
| 11:39.06 | boratynskikamil | Sorry. :-) |
| 11:39.32 | boratynskikamil | undecided: Question. I used this type of recording for incoming connections, right? How to do it for outgoing? Queue too? |
| 11:40.05 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 11:41.31 | undecided | boratynskikamil: incoming and outgoing audio packets are written to separate files |
| 11:41.55 | undecided | so you get both |
| 11:42.20 | undecided | You can use mixmonitor to get them moxed |
| 11:42.26 | undecided | mixed |
| 11:42.49 | boratynskikamil | undecided: My dialplan: |
| 11:43.25 | boratynskikamil | http://wklej.org/id/1283808/ |
| 11:43.33 | *** join/#asterisk D30 (~deo@222.127.13.226) |
| 11:45.10 | boratynskikamil | undecided: As you see, incoming connection are monitored, but not outgoing. |
| 11:46.22 | boratynskikamil | exit |
| 11:46.29 | boratynskikamil | Damn, not here. :-) |
| 11:48.41 | undecided | boratynskikamil: I don't see where your Monitor app is executed, |
| 11:49.11 | undecided | but you can add it to outgoing extensions as well |
| 11:50.14 | boratynskikamil | undecided: Hmm... Not as Queue? |
| 11:50.26 | boratynskikamil | undecided: Incoming is executed in Queue... |
| 11:55.00 | undecided | boratynskikamil: ok, but afair a W or w option is needed for queue recording |
| 11:55.19 | undecided | anyway for outgoing you just use the Monitor application |
| 11:55.24 | undecided | or Mixmonitor |
| 11:57.12 | *** join/#asterisk structz (~andre@194.50.55.200) |
| 11:58.05 | structz | Hi, does anyone have ever experienced something like this: Asterisk MWI on saying you have lots of messages but when going into the voicemail says "you have no messages" |
| 12:08.02 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 12:09.45 | *** join/#asterisk bpietro (~pietro@2-228-138-56.ip191.fastwebnet.it) |
| 12:15.07 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 12:22.41 | bpietro | hi. Is there some timing requirements for REGAUTH packet? For testing purposes I wrote very slow client, sending REGREQ with challenge response (right one, verified), but with timestamp 126 ms, and I got as reply no REGACK nor REGREJ but simple ACK and then server start retransmit REGAUTH packet. |
| 12:27.44 | *** join/#asterisk darkbasic_ (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 12:31.36 | boratynskikamil | undecided: http://wklej.org/id/1283850/ |
| 12:31.38 | boratynskikamil | Like that? |
| 12:37.09 | undecided | boratynskikamil: no |
| 12:37.53 | undecided | I have this |
| 12:37.54 | undecided | exten => 255,1,MixMonitor(file.wav) |
| 12:38.32 | *** join/#asterisk Trixboxer (~Trixboxer@115.124.115.71) |
| 12:38.46 | undecided | no need to use other options |
| 12:39.29 | Trixboxer | Hi, I'm using a Read() command to read user input but if user presses two # then the call gets disconnected |
| 12:39.29 | undecided | and there is no lowercase w option as you have |
| 12:40.22 | undecided | Trixboxer: yes, that's how it's intended to work |
| 12:40.40 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 12:40.45 | Trixboxer | undecided: how can it be avoided ? |
| 12:41.20 | undecided | Trixboxer: do you want to catch the # itself? |
| 12:41.34 | Trixboxer | yeah |
| 12:41.47 | WIMPy | http://voice.yeti.dk/patches/app_read_hash-11.patch |
| 12:41.52 | undecided | then use waitexten |
| 12:41.53 | boratynskikamil | undecided: But.. Wait. I would like to monitor outgoing connection on dahdi/3 channels. |
| 12:41.59 | WIMPy | That will do exactely that. |
| 12:43.05 | undecided | boratynskikamil: I never used dahdi channels . No experience with those |
| 12:44.10 | boratynskikamil | WIMPy: Any suggestion? |
| 12:44.21 | undecided | WIMPy: does that patch work in all asterisk versions? |
| 12:44.52 | WIMPy | That's the patch for Aterisk 11. There's a version for 10 as well. |
| 12:45.17 | undecided | is there for 12? |
| 12:45.18 | WIMPy | boratynskikamil: (mix)monitir doesn't care about channeltypes. |
| 12:45.48 | boratynskikamil | WIMPy: So mine should work properly? |
| 12:45.59 | WIMPy | I haven't tried 12 for quite some time. Try the patch for 11. Chances are that it still fits. |
| 12:46.10 | Trixboxer | undecided: I want to avoid call disconnect due to double press of # |
| 12:47.02 | Trixboxer | one way is to call all # or redirect the call flow by putting exten => #,1,Goto(back) |
| 12:49.43 | undecided | Trixboxer: you should try the patch WIMPy posted |
| 12:50.29 | WIMPy | It's also up on reviewboard. So if someone finds time to do some housekeeping on it, it might become part of the next Asterisk. |
| 12:52.48 | undecided | WIMPy: thank you. I was looking for something like that few months ago |
| 12:53.15 | WIMPy | You should have looked on jira. |
| 12:53.53 | Trixboxer | undecided, WIMPy: thanks. What would be best way to wait for 10 seconds and record whats user speaking |
| 12:54.17 | Trixboxer | I used read here so that user can press # to go to next but double # is dropping call |
| 12:54.20 | WIMPy | For 10s or after 10s? |
| 12:55.05 | Trixboxer | http://pastebin.com/SgwV8KYj |
| 12:55.34 | Trixboxer | here after playing msg1 the system lets user to talk for 15s or till he presses # |
| 12:56.35 | WIMPy | Maybe you should just use Record instead then? |
| 12:56.59 | WIMPy | And the combination of Background and Read does not look very sensible. |
| 12:57.17 | WIMPy | Did you notice that Read can also play sounds? |
| 12:58.20 | boratynskikamil | WIMPy: http://wklej.org/id/1283881/ |
| 12:58.25 | boratynskikamil | Repaired extension. |
| 12:58.39 | boratynskikamil | If I hangup the call, at the beginng, I mean I do not receive it. |
| 12:58.55 | boratynskikamil | I see in logs that MixMonitor has been started and ended. |
| 12:59.02 | boratynskikamil | If I receive, I do not see MixMonitor at all. |
| 12:59.30 | WIMPy | You start Mixmonitor after the call has failed. |
| 13:00.18 | WIMPy | If you want to record the call, you obviousely have to start MixMonitor before you send the call to the destination. |
| 13:00.48 | *** join/#asterisk jploh (~textual@49.146.167.241) |
| 13:01.17 | Trixboxer | WIMPy: I have started MixMonitor before Background(msg1) |
| 13:01.30 | WIMPy | That was for boratynskikamil |
| 13:01.35 | Trixboxer | so that it records both, whats being played and what user is saying |
| 13:02.28 | WIMPy | (And dahdi/3 does not look like OUT_ISDN) |
| 13:03.03 | boratynskikamil | WIMPy: Repaired. |
| 13:03.17 | boratynskikamil | It was filename, sorry. :-) |
| 13:06.34 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 13:11.52 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 13:16.06 | *** join/#asterisk ChaosPsyke (~Psyke@41.77.102.228) |
| 13:20.15 | ChaosPsyke | Hi guys, I am having an issue with the "leastrecent" ring strategy. When a client calls, asterisk plays IVR and then goes to least recent phone. If it rings for more than 10 seconds it timesout out and goes back to asterisk, asterisk plays IVR again but sends the call back to the extension which did not pickup. Basically locking the call in an infinite loop. Has anyone else encountered this before? |
| 13:23.29 | *** join/#asterisk calum_ (~calum_@host86-171-67-5.range86-171.btcentralplus.com) |
| 13:48.48 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 13:57.47 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 14:04.44 | *** join/#asterisk Katty (~Katty@97-91-121-70.static.stls.mo.charter.com) |
| 14:08.30 | Katty | POST IT NOTE. |
| 14:10.05 | MaliutaLap | On the forehead? |
| 14:10.17 | MaliutaLap | you're Sherlock Holmes! |
| 14:10.19 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 14:11.49 | *** join/#asterisk serafie (~erin@nat/digium/x-fjsdhexilokkystn) |
| 14:15.32 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:19.15 | *** join/#asterisk Blashyrkh (~michaeld@94.136.25.97) |
| 14:30.34 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:30.34 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:30.44 | *** join/#asterisk xaristax (~xaristax@148.240.65.35) |
| 14:30.58 | xaristax | Hi i wonder how can i make to after a dial create a conference? |
| 14:31.32 | xaristax | i already do this but withot success Dial(SIP/blbla,300,U(conf),) |
| 14:32.27 | *** join/#asterisk theron (~theron@69.63.185.56) |
| 14:36.38 | *** join/#asterisk Milarepa_ (~Milarepa@host-74-211-92-125.beyondbb.com) |
| 14:46.00 | *** join/#asterisk Neozonz (~arajakul@unaffiliated/neozonz) |
| 14:46.08 | Neozonz | Do i need the |
| 14:46.08 | Neozonz | extensions.ael file? |
| 14:50.36 | *** join/#asterisk maruen (~maruen@179.99.252.27) |
| 14:51.03 | maruen | Hi folks, does anyone could help me install asterisk in mac os Mavericks? |
| 14:51.10 | maruen | I could install it in Ubuntu |
| 14:51.16 | maruen | but not in Mac OS X |
| 14:52.40 | Neozonz | hmm |
| 14:52.56 | Neozonz | so i created a context instead extensions but when i do dialpan show default |
| 14:52.59 | Neozonz | nothing... |
| 14:53.05 | Neozonz | i also reloaded dialpan |
| 14:53.11 | Neozonz | and restarted asterisk no go... |
| 14:53.43 | [TK]D-Fender | [09:46]Neozonzextensions.ael file? <- no |
| 14:53.48 | Penguin | "Need" is a subjective term. Only you know if you NEED ael. |
| 14:54.17 | [TK]D-Fender | [09:52]Neozonzso i created a context instead extensions but when i do dialpan show default <- editing the wrong file, or wrong permissions |
| 14:55.02 | maruen | Actually I installed on Mac OS X, but I cant run it, it gets this error: chan_iax2.c:9260 timing_read: Timer failed acknowledge |
| 14:55.02 | maruen | Assertion failed: (mod_evsub.mod.id != -1), function pjsip_evsub_register_pkg, file ../src/pjsip-simple/evsub.c, line 415 |
| 14:55.18 | maruen | Anybody knows something about it? |
| 14:56.23 | Neozonz | -rw-r----- 1 asterisk asterisk 307 Feb 26 14:35 extensions.conf |
| 14:57.13 | Penguin | Is asterisk set to run as user asterisk and group asterisk? |
| 14:57.44 | Neozonz | asterisk 19052 1 0 14:55 ? 00:00:01 /usr/sbin/asterisk -p -U asterisk |
| 14:58.04 | Penguin | And the ownership/permissions on the directories above that file? |
| 14:58.16 | [TK]D-Fender | Show us this "no go" |
| 14:58.33 | [TK]D-Fender | and your extensions.conf |
| 14:58.39 | Neozonz | drwxr-xr-x 4 asterisk asterisk 4096 Feb 26 14:55 asterisk |
| 14:58.51 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
| 14:59.06 | Neozonz | http://prntscr.com/2w0j34 |
| 14:59.55 | Neozonz | http://prntscr.com/2w0jav |
| 15:00.03 | Penguin | Someone didn't read the book. |
| 15:01.28 | Penguin | Did you run "dialplan reload" after you saved changes to the file? |
| 15:01.50 | Neozonz | yes... |
| 15:01.52 | Penguin | Was there any output from running the command? |
| 15:02.00 | Neozonz | my screenshot |
| 15:02.08 | Neozonz | but none of the things in extensions.conf |
| 15:02.10 | Penguin | You didn't show that. |
| 15:02.28 | Neozonz | http://prntscr.com/2w0jav |
| 15:02.33 | Penguin | You didn't show that. |
| 15:02.41 | *** join/#asterisk MauriceM_ (~MauriceM_@66-193-40-64.static.twtelecom.net) |
| 15:03.04 | Neozonz | nm i fixed it |
| 15:03.07 | Neozonz | i was missing priority |
| 15:03.12 | Neozonz | but with it missing there are no errors |
| 15:03.34 | Penguin | dialplan reload didn't show you ANY information about your mistake? |
| 15:03.54 | Penguin | tests |
| 15:04.49 | Penguin | cpe-e650*CLI> dialplan reload |
| 15:04.49 | Penguin | Dialplan reloaded. |
| 15:04.49 | Penguin | [Feb 26 09:04:39] NOTICE[31678]: pbx.c:4514 pbx_extension_helper: Cannot find extension '7000' in context '' |
| 15:04.52 | Penguin | [Feb 26 09:04:39] WARNING[31678]: pbx_config.c:1524 pbx_load_config: Invalid priority/label 'Dial' at line 48 of extensions.conf |
| 15:04.56 | Penguin | ^ A CLUE |
| 15:05.03 | Neozonz | ? |
| 15:05.06 | Neozonz | i didnt see that |
| 15:05.37 | Penguin | You probably didn't run dialplan reload. |
| 15:05.43 | Penguin | You certainly didn't show me that you ran it. |
| 15:07.21 | Neozonz | I lied |
| 15:07.25 | Neozonz | I didn't notice it |
| 15:07.27 | Neozonz | sorry |
| 15:08.24 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:08.25 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:12.13 | *** part/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 15:13.00 | Neozonz | How would my dialpan look if I configure my users by name? |
| 15:13.11 | Penguin | normal |
| 15:13.24 | Neozonz | it will still dial by extenion? |
| 15:13.35 | Penguin | Extensions execute applications. |
| 15:13.59 | Penguin | You will always enter extensions on phones to make calls. |
| 15:14.25 | Penguin | Then the extensions execute whatever applications you configure them to execute. |
| 15:14.55 | Penguin | To put the call to another phone, use the Dial() application. |
| 15:15.10 | Neozonz | Cool, thanks |
| 15:15.19 | Neozonz | I think I know what to do now |
| 15:15.21 | Neozonz | appreciate the help! |
| 15:16.31 | Penguin | The Dial() application accepts the syntax of channel_tech/device_name,timeout,options. E.g., Dial(SIP/001122334455,26) |
| 15:16.56 | Penguin | SIP channel tech, device named 001122334455, timeout of 26 seconds, with no additional options |
| 15:17.23 | Penguin | This is all in The Book, by the way. |
| 15:17.27 | Penguin | ~book |
| 15:17.27 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 15:18.25 | Neozonz | I've been reading it but its small syntax mistakes that are killing me |
| 15:18.41 | Neozonz | for example, ldap does not want to find the exten... |
| 15:19.14 | Penguin | I don't think you put extensions in ldap. |
| 15:19.29 | Neozonz | I did |
| 15:19.36 | Neozonz | i'm looking at ldap browser |
| 15:19.43 | Neozonz | and i have both the objectclass and extensions |
| 15:19.49 | Penguin | I'm not so sure asterisk works like that. |
| 15:19.59 | Neozonz | ? |
| 15:20.56 | Penguin | LDAP is a directory protocol. I don't know how you could possibly make asterisk execute extensions there. |
| 15:22.11 | Neozonz | it's pulling extensions + caller id from ldap |
| 15:22.18 | Neozonz | then does the calls via asterisk |
| 15:22.22 | Neozonz | using the realtime ldap module |
| 15:22.38 | Penguin | I'm probably just not understanding what you're doing. |
| 15:23.12 | Penguin | I don't use LDAP, so I don't know how to integrate it with the dialplan. |
| 15:26.45 | *** join/#asterisk navaismo (~navaismo@201.124.146.234) |
| 15:33.05 | *** join/#asterisk wonderworld (~ww@ip-62-143-158-113.unitymediagroup.de) |
| 15:33.13 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 15:35.46 | Neozonz | how does one setup realtime dialplans? |
| 15:37.00 | navaismo | in the db |
| 15:37.36 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 15:38.41 | Neozonz | what if it was ldap |
| 15:38.59 | navaismo | idk |
| 15:39.52 | Neozonz | for example |
| 15:39.53 | Neozonz | http://free.oxymium.net/Asterisk/README.realtime-ldap |
| 15:40.03 | Neozonz | [default] |
| 15:40.03 | Neozonz | switch => Realtime/default@realtime_ext |
| 15:40.06 | Neozonz | that's it?? |
| 15:43.28 | navaismo | never tried that |
| 15:44.41 | Neozonz | i've set it all up |
| 15:44.46 | Neozonz | but it still says extension missing :( |
| 15:47.29 | *** join/#asterisk Farkie (~Farkie@213.229.88.2) |
| 15:49.03 | Farkie | Hey, having a strange issue. If we have a incoming caller (A), it goes to a group. now if someone (B) answers and tries to transfer internally to another colleague (C), it's silent when doing assisted transfer from B -> C, but if B hangs up, A can speak to C.. any ideas / more info I can give? |
| 15:49.09 | file | I wonder why they wrote that... |
| 15:50.16 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 15:51.50 | Neozonz | [TK]D-Fender, http://pastebin.com/rSujWpaE |
| 15:52.02 | Neozonz | any help would be most appreciated, getting closer with ldap stuff |
| 15:52.14 | Neozonz | its grabbing the extensions just when i make calls it breaks now :( |
| 15:52.32 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-uvesyyijpqydlnhe) |
| 15:52.32 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 15:52.51 | [TK]D-Fender | Neozonz: You keep asking for help and you aren't showing us what it's supposed to be pulling |
| 15:53.50 | Neozonz | it's pulled the extention correctly now |
| 15:53.56 | Neozonz | issue is calling the extention |
| 15:54.08 | [TK]D-Fender | DEVICE |
| 15:54.32 | [TK]D-Fender | We don;t see that you even have an entry for it. |
| 15:55.00 | *** join/#asterisk MauriceM_ (~MauriceM_@66-193-40-64.static.twtelecom.net) |
| 15:55.10 | Neozonz | device? |
| 15:57.17 | *** join/#asterisk caveat- (hoax@2a01:4f8:191:9111:30::10) |
| 15:57.58 | [TK]D-Fender | I see a DIAl that is not going through... |
| 15:58.03 | Neozonz | ok... so i had to modify my extentions for ldap to work |
| 15:58.18 | [TK]D-Fender | You just shoved a pastebin and didn't tell us what you thought was wrong with it |
| 15:58.25 | [TK]D-Fender | Your descriptions are very very weak |
| 15:58.26 | Neozonz | exten => _X.,1,Dial(SIP/arajakulasingam) |
| 15:58.54 | Neozonz | how do i replace arajakulasingam with the username? or the [nameofheading] in sip.conf |
| 15:59.14 | [TK]D-Fender | that IS the name |
| 15:59.17 | Neozonz | i know u can dial by ${EXTEN} but in this case its not an extension but rather a name |
| 15:59.38 | Neozonz | is there something like ${EXTEN} or ${NAME} |
| 15:59.51 | [TK]D-Fender | [10:58]Neozonzexten => _X.,1,Dial(SIP/arajakulasingam) <-- there is no magical association of a text name to a VARIABLE number |
| 16:00.05 | [TK]D-Fender | Three is nothing that says "john is 1000" |
| 16:00.13 | [TK]D-Fender | How is Asterisk supposed to know this? |
| 16:00.17 | [TK]D-Fender | this is YOUR job to set up |
| 16:01.06 | [TK]D-Fender | So get ready to make dialplan lines for EACH of your numbers specifying the device to be called |
| 16:01.21 | *** join/#asterisk yago (~kresp0@gateway/tor-sasl/kresp0) |
| 16:02.50 | MauriceM_ | Quick question on Asterisk IVR over lets say, Zanzibar, or Voxeo Prophecy. Anyone using Asterisk PBX with a different IVR solution? |
| 16:05.55 | Katty | looks in |
| 16:06.11 | [TK]D-Fender | looks out |
| 16:06.17 | Kobaz | looks around |
| 16:06.53 | [TK]D-Fender | bases = covered |
| 16:07.33 | Katty | so many prepositional phrases :> |
| 16:08.28 | file | wobbles |
| 16:12.10 | Penguin | neozonz: That "heading" as you put it is the NAME of the DEVICE. The characters between the square brackets make the "username" that the device (often a phone) must use for authentication to asterisk and it is also the name of the device (a phone) that you Dial() when you want to put a call to the device (phone). |
| 16:12.46 | Blashyrkh | <PROTECTED> |
| 16:13.07 | Blashyrkh | iaxmodem and asterisk are on the same host |
| 16:13.09 | Neozonz | how can i reference device NAME from dialpan via an argument like ${EXTEN} is that possible? |
| 16:13.45 | [TK]D-Fender | Neozonz: ${exten} is what you DIALED. the pattern you used is numeric. |
| 16:14.06 | [TK]D-Fender | Neozonz: Where are you expecting asterisk to match a list of numbers to names from? |
| 16:14.14 | Neozonz | from users.conf |
| 16:14.20 | [TK]D-Fender | [11:00][TK]D-FenderThree is nothing that says "john is 1000" <---- |
| 16:14.23 | [TK]D-Fender | Neozonz: No. |
| 16:14.27 | Neozonz | and/or the realtime ldap |
| 16:14.41 | [TK]D-Fender | Neozonz: The dialplan knows nothing of any kind of association in users.conf |
| 16:14.42 | Neozonz | whats the point of realtime ldap if it doesnt match users to extentions.. |
| 16:14.53 | Penguin | People utilize the users.conf file? |
| 16:15.00 | [TK]D-Fender | Neozonz: and you should never ever ever be using that file no matter what. |
| 16:15.12 | Neozonz | I'm trying to lol |
| 16:15.16 | [TK]D-Fender | STOP NOW |
| 16:15.21 | Neozonz | yes sir |
| 16:15.38 | Penguin | Since LDAP is a directory protocol, that would be a perfect place to say "John uses extension 1000." |
| 16:15.42 | [TK]D-Fender | sip.conf , iax.con, etc. Use the proper individual bits |
| 16:15.56 | *** join/#asterisk boratynskikamil (~kamilbora@109.231.38.149) |
| 16:15.59 | Neozonz | Penguin, i ahve the mappings done for ldap |
| 16:15.59 | Penguin | Then when you call 1000, asterisk can Dial(SIP/johns_phone). |
| 16:16.18 | Neozonz | but dialpan doesn't seem to associate johns_phone with extention frmo ldap |
| 16:16.44 | [TK]D-Fender | Neozonz: Of course not... you put what you did in Dial(). |
| 16:16.55 | [TK]D-Fender | Neozonz: there is no assiation. That exists only in your head |
| 16:17.06 | [TK]D-Fender | Neozonz: If you want a lookup you have to code it in the dialplan. |
| 16:17.07 | Neozonz | exten => _X.,1,Dial(SIP/${EXTEN}) |
| 16:17.07 | Neozonz | exten => _X.,n,Hangup() |
| 16:17.38 | [TK]D-Fender | [11:17]Neozonzexten => _X.,1,Dial(SIP/${EXTEN}) <- this takes the NUMBER you dialed and shoves it directly in the dial. You did not do any kind of lookup |
| 16:17.55 | [TK]D-Fender | Neozonz: Dialplan = programming. |
| 16:17.56 | Neozonz | how does one take the number i dialed, look it up against realtime ldap |
| 16:17.59 | boratynskikamil | http://wklej.org/id/1284056/ |
| 16:18.02 | boratynskikamil | Question, if I may. |
| 16:18.10 | [TK]D-Fender | Neozonz: YOU have to do a few steps PRIORI to dialing to do this lookup yourself |
| 16:18.16 | boratynskikamil | Why outgoing calls are not hanguped by default? |
| 16:18.29 | boratynskikamil | I mean, softphone freezes. |
| 16:19.25 | [TK]D-Fender | boratynskikamil: What dsoftphone? We see dialplan ... and a DAHDI channel that should get dialed. fix your description and show us a complete scenio including the output of the failure |
| 16:19.36 | Penguin | neozonz: Store the looked up data into a variable, possibly ${DEVICE}, and then Dial(SIP/${DEVICE}). |
| 16:19.53 | Penguin | That's how I do it using the astDB. |
| 16:20.09 | [TK]D-Fender | Penguin: give or take the actual DB funtion calls :) |
| 16:20.28 | Penguin | The lookup is performed before the Dial(), of course. |
| 16:21.18 | Penguin | But I guess you could do the lookup right there inside the Dial() application using the same functions used to do the lookup earlier. |
| 16:21.32 | boratynskikamil | [TK]D-Fender: QuteCom. |
| 16:21.32 | boratynskikamil | In fact, there is no "failure". It just freezes. |
| 16:22.09 | [TK]D-Fender | boratynskikamil: We see nothing about a "softphone" in that pastebin. |
| 16:22.12 | Neozonz | Penguin, I think I'm already storing it http://prntscr.com/2w17fs |
| 16:22.22 | boratynskikamil | I mean, I have to click hangup to close connection in QuteCom. |
| 16:22.24 | [TK]D-Fender | Neozonz: You are RETRIEVING it. |
| 16:22.28 | [TK]D-Fender | aren't* |
| 16:23.16 | Katty | tangles [TK]D-Fender in a ball of yarn. |
| 16:23.16 | Penguin | I don't see anything related to storing device name related to the extension used to dial said device. |
| 16:23.56 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 16:24.12 | Neozonz | I don't understand... |
| 16:24.15 | Neozonz | but will keep at it |
| 16:26.44 | [TK]D-Fender | Neozonz: YOu have to do a database lokup in extensions.conf <------------- |
| 16:27.06 | [TK]D-Fender | Neozonz: Your dialplan does exactly what you tell it to. You didn't tell it to look anything up |
| 16:27.59 | [TK]D-Fender | Neozonz: there is no "association". There is only "programming? You define the steps, and this include database lookups when you expect to take "A'" and look it up somewhere and come back with "B" |
| 16:28.11 | Neozonz | I dont understand then |
| 16:28.16 | Neozonz | what the heck is the point of res_ldap.conf |
| 16:28.21 | Neozonz | does it not store the values? |
| 16:28.29 | [TK]D-Fender | YOU have to LOOK THEM UP |
| 16:28.30 | [TK]D-Fender | ^ |
| 16:28.45 | Neozonz | [sip] |
| 16:28.45 | Neozonz | name = uid |
| 16:28.48 | [TK]D-Fender | no |
| 16:28.59 | [TK]D-Fender | doesn't matter what's in that DB, or that config file |
| 16:29.10 | [TK]D-Fender | exten => _X.,1,Dial(SIP/${EXTEN}) <--- these are the STEPS your call takes as it processes.... |
| 16:29.16 | Neozonz | ;exten => 7002,1,Dial(SIP/${NAME}/${EXTEN}) |
| 16:29.18 | Neozonz | i tried that |
| 16:29.21 | [TK]D-Fender | You dnd't take a step to do a lookup |
| 16:29.38 | [TK]D-Fender | ${NAME} is not a LOOKUP command |
| 16:29.48 | Neozonz | where can I find a list of lookup commands |
| 16:29.52 | [TK]D-Fender | you are inventing syntax and thinging that extensions.conf knows ANYTHING about LDAP |
| 16:29.57 | [TK]D-Fender | it DOESN'T |
| 16:30.06 | [TK]D-Fender | You have to do a FUNCTION call to pull those values |
| 16:30.38 | *** part/#asterisk Trixboxer (~Trixboxer@115.124.115.71) |
| 16:30.47 | Neozonz | is there any guide or read me regarding doing that |
| 16:31.05 | Penguin | It's not in the book? |
| 16:31.08 | Neozonz | nothing in any of the guides say anything about that |
| 16:31.11 | [TK]D-Fender | "core show applications" |
| 16:31.15 | [TK]D-Fender | "core show functions" |
| 16:31.19 | [TK]D-Fender | and go read the BOOK. |
| 16:31.46 | Neozonz | https://wiki.asterisk.org/wiki/display/AST/LDAP+Realtime+Driver |
| 16:31.49 | file | nothing specific to LDAP |
| 16:31.51 | Neozonz | nothing |
| 16:31.53 | [TK]D-Fender | There is an entire chapter on "database integration", and getting information from databases is part of it, |
| 16:31.55 | Penguin | I assume LDAP is just another relational database. |
| 16:32.02 | [TK]D-Fender | neothat is not the book |
| 16:32.03 | [TK]D-Fender | ~boot |
| 16:32.04 | infobot | boot is, like, what you get when you act like a EFNet user, or #debian-boot |
| 16:32.08 | [TK]D-Fender | ~book |
| 16:32.08 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:32.10 | [TK]D-Fender | ^^^^^^^^ |
| 16:32.50 | Neozonz | http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#ExternalServices_id291590 |
| 16:32.53 | Neozonz | nothing... |
| 16:32.57 | Penguin | If it's just another relational database, then you store a primary key and associate other data with that key. If the key is NAME, then the associated data could be EXTENSION. |
| 16:32.59 | [TK]D-Fender | 4th edition there |
| 16:33.23 | [TK]D-Fender | [11:32]Neozonznothing... <- do not expect the word "LDAP". *'s databse integration is GENERIC |
| 16:33.24 | Penguin | So then you do a lookup in the DB. |
| 16:33.59 | [TK]D-Fender | Neozonz: this is NOT LDAP specific |
| 16:34.10 | Penguin | If you lookup a NAME of John in the DB and there is an EXTENSION associated with it, then that's the extension to execute. |
| 16:34.31 | [TK]D-Fender | [11:32]Neozonzhttp://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#ExternalServices_id291590 <- chapter 16 |
| 16:34.40 | Penguin | If there is a primary key of EXTENSION and you need to relate a DEVICE to it, then add that data too. |
| 16:35.22 | Penguin | If you put a call to extension 1000, do a DB lookup to see what extension 1000 relates to. Maybe it is a DEVICE of 00001111FFFF. |
| 16:35.41 | Penguin | Then you go back to the dialplan and execute Dial(SIP/00001111FFFF). |
| 16:36.34 | Penguin | ANY and ALL associations between people, the devices people use, and the extensions used to dial those devices must be abstracted by you the programmer. |
| 16:37.56 | Penguin | Any problem with this concept? |
| 16:39.34 | Penguin | Depending on the style of the relational database, you might be able to put all the various associated pieces together in one table. |
| 16:45.14 | *** join/#asterisk Neoti (~Thunderbi@cpc5-nott16-2-0-cust33.12-2.cable.virginm.net) |
| 16:59.33 | Neozonz | so whats the point of all that stuff in res_ldap.conf |
| 17:07.08 | file | looks like it maps fields within LDAP to internal Asterisk configuration options |
| 17:07.23 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:10.30 | *** join/#asterisk tuxd00d (~tuxd00d@ip24-251-34-116.ph.ph.cox.net) |
| 17:12.40 | *** join/#asterisk cmendes0101 (~cmendes01@office.phone.com) |
| 17:14.42 | *** join/#asterisk Neoti (~Thunderbi@cpc5-nott16-2-0-cust33.12-2.cable.virginm.net) |
| 17:14.55 | *** part/#asterisk Blashyrkh (~michaeld@94.136.25.97) |
| 17:16.22 | Neozonz | then couldn't i lookup values against the internal asterisk config options? |
| 17:16.49 | Neozonz | how exactly does it map it... |
| 17:16.57 | Neozonz | because i seem to be missing the peer name for any entry it maps |
| 17:32.29 | boratynskikamil | Question, is it possible to specify notification for call? I mean, if connection comes from CARD1 it shows number + [CARD1]? |
| 17:37.23 | navaismo | override the callerid name |
| 17:37.39 | boratynskikamil | navaismo: As I thought. Thanks a lot. |
| 17:40.36 | *** join/#asterisk Warp4 (~Warp4@mailfrom.esg.md) |
| 17:40.45 | Warp4 | man i havent been in this channel in quite some time |
| 17:41.48 | *** join/#asterisk aynam (5896ca8a@gateway/web/freenode/ip.88.150.202.138) |
| 17:42.08 | Warp4 | hi all. I have an issue with digium switchvox where when employees are termed and new phone extensions are put into the system to replace them, call reporting is not working with the new extensions. However, with the legacy extensions the call reporting is working fine. |
| 17:43.47 | Warp4 | Basically wanting to see which numbers are going to what extension when searched by caller ID. To reiterate, this is working fine with older extension put in place when the phone system was installed, but not when new extensions are added. |
| 17:48.05 | [TK]D-Fender | [12:16]Neozonzthen couldn't i lookup values against the internal asterisk config options? <- you could... you have to DO this a a DIALPLAN STEP. |
| 17:48.32 | Neozonz | any examples anywhere regarding this? ^ |
| 17:48.43 | [TK]D-Fender | Warp4: Switchvox is unable to be supported here as it is a closed solution. |
| 17:48.51 | [TK]D-Fender | Neozonz: Already given to you |
| 17:49.07 | [TK]D-Fender | Neozonz: the BOOK, and the application & function lists. |
| 17:51.11 | Penguin | I'll give you an example using the built-in asterisk DB. |
| 17:52.07 | Penguin | same => n,Set(DEVICE=${EVAL(${DB(phones/${EXTEN}/device)})}); |
| 17:52.58 | Penguin | That performs the lookup in the DB. The value in the DB is set to variable ${DEVICE}. |
| 17:53.18 | Penguin | same => n,Dial(${DEVICE},26,rx); |
| 17:53.27 | Penguin | That performs the Dial to the device. |
| 17:54.07 | Penguin | This has nothing to do with LDAP, though, so don't think you can paste my code into your dial plan and make it to LDAP lookups. |
| 17:54.17 | [TK]D-Fender | that is for AstDB. that isn't a lookup into the one he wants to use |
| 17:54.51 | *** join/#asterisk koffel (koffel@173-167-212-107-ip-static.hfc.comcastbusiness.net) |
| 17:55.35 | Penguin | Fully aware. |
| 17:55.38 | Penguin | (1151.11) <Penguin> I'll give you an example using the built-in asterisk DB. |
| 17:56.23 | koffel | anyone here can help me with asterisk 1.8 and a polycom ip 320 phone? |
| 17:56.49 | koffel | i can not get this phone to register to asterisk no matter what i do |
| 17:57.34 | Penguin | I already disclaimed knowing how to integrate LDAP into asterisk dialplan. Concepts I'm okay with, my doing it is a different story. |
| 17:58.10 | *** join/#asterisk calum_ (~calum_@cpc4-harg5-2-0-cust371.7-1.cable.virginm.net) |
| 18:03.31 | *** join/#asterisk Milarepa (~Milarepa@host-74-211-92-125.beyondbb.com) |
| 18:04.38 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 18:06.51 | *** join/#asterisk yago (~kresp0@gateway/tor-sasl/kresp0) |
| 18:15.27 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 18:20.23 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 18:22.49 | boratynskikamil | Question. What if I would like to store recordings in directory tree incoming/year-month/day? Will Asterisk create a directory tre on his on or I should execute my own, written script? |
| 18:29.08 | [TK]D-Fender | What recordings? Made how? |
| 18:30.29 | *** join/#asterisk elcontrastador (~textual@12.226.100.130) |
| 18:30.31 | protocoldoug | With MixMonitor() it will create the directory at least in my test with 11.6-certified |
| 18:30.35 | protocoldoug | boratynskikamil: ^^ |
| 18:31.03 | protocoldoug | I did a MixMonitor(/tmp/directorydidntexist/foo.ulaw) and it created the dir /tmp/directorydidntexist |
| 18:33.33 | *** join/#asterisk ghost75 (~quassel@dslb-088-066-162-144.pools.arcor-ip.net) |
| 18:34.30 | boratynskikamil | [TK]D-Fender: MixMonitor. |
| 18:34.49 | boratynskikamil | Nice. |
| 18:35.48 | Penguin | protocoldoug: What if you have several directories deep that do not exist? /tmp/another/directory/that/doesnt/exist/foo.ulaw |
| 18:36.55 | elcontrastador | I'm trying to get pickup groups working with no luck. I'd like x3980 to be able to pickup x3910 and x3920. Can someone see anything wrong here? http://pastebin.com/mpDmZwxc |
| 18:37.21 | boratynskikamil | Penguin: Interesting question. Is it any suggestion? Won't it work properly? |
| 18:37.33 | *** join/#asterisk wonderworld (~ww@ip-62-143-58-183.unitymediagroup.de) |
| 18:41.25 | Penguin | elcontrastador: I don't know what that channel setting is for, but phones ces-admin-3910, ces-admin-3920, and ces-admin-3980 are all in the same callgroup and pickup group. That means that a call to any of the three devices can be picked up by any of the other devices. |
| 18:42.50 | elcontrastador | The channel settings are out of desperation! :-) I understood that this should be working with just the shown sip config and pickupexten shown in the features.conf. It is not. |
| 18:43.25 | elcontrastador | Asterisk 11.6.0 |
| 18:43.36 | Penguin | Adding non-existent settings to phone entries won't make it start working. Is that channel setting something new in 11? |
| 18:44.03 | elcontrastador | https://wiki.asterisk.org/wiki/display/AST/Call+Pickup |
| 18:45.04 | *** join/#asterisk bkruse (~Adium@24.42.229.8) |
| 18:45.09 | Penguin | I don't see anything there about adding a channel setting to a device in sip.conf. |
| 18:46.26 | navaismo | koffel, what is the issue, did you see in the cli the registration attempts |
| 18:49.05 | elcontrastador | Penguin: I read that channel setting somewhere. I removed it, reloaded sip, and can't pickup but will play the beeperror. Should the sip entries callgroup and pickupgroup be all that's necessary for this to work? |
| 18:49.42 | *** join/#asterisk jansiva (~janaki@118.102.128.225) |
| 18:51.15 | Penguin | The callgroup to set the group that the device is in, the pickupgroup to set which groups a device can pickup, and either the Pickup() application or pickup feature... |
| 18:52.11 | elcontrastador | The pickupexten feature shown should work the way it's configured then. |
| 18:52.18 | *** join/#asterisk gaul1 (~gaul1@e180046023.adsl.alicedsl.de) |
| 18:52.37 | Penguin | Did you fully restart asterisk after making changes to features.conf? |
| 18:53.01 | elcontrastador | no just did 'reload features' |
| 18:53.41 | Penguin | If possible, restart asterisk fully and check for new results. |
| 18:54.33 | elcontrastador | screw 'em...just restarted it :-) I've gotta get this working. |
| 18:54.44 | Penguin | haha |
| 18:55.06 | Penguin | I like core restart gracefully for times like that. |
| 18:55.16 | elcontrastador | same results |
| 18:55.40 | Penguin | I guess it's time to look at the debug. |
| 18:55.52 | elcontrastador | ok...1 sec |
| 18:56.22 | elcontrastador | let me pipe output to filter out all this damn dpma crap |
| 18:57.14 | Penguin | I used to use pickup groups in 1.4, but I don't know if I ever did in 1.8, and I haven't used 11 yet. |
| 19:01.09 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 19:02.43 | elcontrastador | http://pastebin.com/2jPYNgNk |
| 19:03.52 | elcontrastador | I don't see much there...debug 7, verbose 7 |
| 19:06.18 | Penguin | I don't see where you dialed *8 to pickup any ringing channels. |
| 19:09.05 | elcontrastador | called from outside line, picked up 3980 and dialed *8, played beeperr (shown in log) |
| 19:10.05 | elcontrastador | line 41 only thing that shows anything related to pickup as far as I can see |
| 19:10.11 | Penguin | I do see where that file played. Are you willing to try using the other method to pickup? |
| 19:10.23 | elcontrastador | yes, anything |
| 19:10.47 | *** join/#asterisk wonderworld (~ww@ip-62-143-58-183.unitymediagroup.de) |
| 19:11.03 | Penguin | Disable the pickup stuff in features.conf and add a new extension to run the pickup apps. |
| 19:12.03 | elcontrastador | exten => *8,1,Pickup(3910) ? |
| 19:12.42 | elcontrastador | I've tried this with no extension...should work to pickup calls in your pickupgroup and didn't work |
| 19:12.57 | Penguin | I wouldn't do that. I would base it on channels not extensions. |
| 19:13.40 | elcontrastador | could you please give me an example? |
| 19:14.44 | Penguin | You can use the Pickup() application to pick up channels based on a special variable set to channels, or you can use PickupChan() to pick up based on the device. |
| 19:15.22 | *** join/#asterisk tonyclewis (uid6025@gateway/web/irccloud.com/x-njhbpzcxkuewfkxg) |
| 19:16.46 | Penguin | If you use PickupChan(), you'll have to programmatically determine which device's channel to pick up prior to execution of the application. |
| 19:17.23 | Penguin | But after looking at the Pickup() application, you may not need to specify any data at all for it. Try *8,1,Pickup() and nothing else in it. |
| 19:17.56 | Penguin | If I am reading this correctly, Pickup() will use the group you set for the devices in sip.conf. |
| 19:18.11 | elcontrastador | yes, that's what i thought |
| 19:18.14 | elcontrastador | i will try again now |
| 19:18.22 | Penguin | 1) If no <extension> targets are specified, the application will pickup a |
| 19:18.22 | Penguin | channel matching the pickup group of the requesting channel. |
| 19:19.05 | elcontrastador | fast busy |
| 19:19.06 | elcontrastador | for Notify User ces-admin-3910 (queued) |
| 19:19.06 | elcontrastador | <PROTECTED> |
| 19:19.06 | elcontrastador | <PROTECTED> |
| 19:19.11 | elcontrastador | oops...sorry |
| 19:19.21 | Penguin | you're okay |
| 19:19.42 | Penguin | So your other devices are set to group 1. If your phone is in pickupgroup 1, it should pick up a call to any other device in group 1. |
| 19:21.25 | Penguin | Option 2 is the way I mentioned already, with the special variable. |
| 19:22.15 | Penguin | You can Set(PICKUPMARK=${EXTEN}) before the Dial() in that extension that is ringing the two phones. |
| 19:24.41 | Penguin | If that was extension 3000, you can make *8 execute Pickup(3000@PICKUPMARK). |
| 19:25.34 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 19:25.36 | Penguin | Oh, you probably didn't try Pickup(3000) earlier. |
| 19:26.13 | elcontrastador | no, i did not try pickup 3000 |
| 19:26.31 | Penguin | I didn't see anything execute extension 3910. |
| 19:26.42 | Penguin | That's why Pickup(3910) wouldn't do anything. |
| 19:27.13 | Penguin | Extension 3910 probably doesn't exist (from an active channel standpoint). |
| 19:27.34 | elcontrastador | tried Pickup(3000)...fast busy |
| 19:28.04 | Penguin | Try setting the pickup mark in extension 3000 right before the Dial(). Then Pickup(3000@PICKUPMARK). |
| 19:28.05 | *** join/#asterisk darkdrgn2k (~darkdrgn2@209.90.253.66) |
| 19:28.07 | darkdrgn2k | Hi all |
| 19:28.16 | elcontrastador | y...doing that now |
| 19:28.18 | Penguin | After that, I'm out of options. |
| 19:28.28 | darkdrgn2k | any one know if there is a way to stablizie the clock drift in esxi ? |
| 19:28.30 | Penguin | Those are the only ways I know how to pick up. |
| 19:28.48 | darkdrgn2k | voipmonitor is doing wierd thing when crunching sip data |
| 19:28.56 | elcontrastador | damn...same thing |
| 19:29.18 | Penguin | Something's missing. I'm not seeing why this failure exists. |
| 19:29.28 | Chainsaw | darkdrgn2k: Sync against NTP servers on the ESXi hosts. |
| 19:29.54 | elcontrastador | you'd think they're be more in the logs |
| 19:30.52 | darkdrgn2k | <Chainsaw: i dont mean CLOCK CLOCK i mean <Chainsaw> |
| 19:31.07 | darkdrgn2k | or /sys/devices/system/clocksource/clocksource0/current_clocksource even |
| 19:31.36 | Chainsaw | Syntax error. |
| 19:34.13 | darkdrgn2k | Chainsaw: i dont mean CLOCK CLOCK i mean /sys/devices/system/clocksource/clocksource0/current_clocksource |
| 19:34.13 | darkdrgn2k | LOL |
| 19:34.57 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 19:34.59 | Chainsaw | darkdrgn2k: It remains important to NTP-sync the ESXi host. If you do not, expect the VMs to drift like nobodies business. |
| 19:37.16 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 19:38.11 | elcontrastador | Penguin: changed exten to *88 and got it working with Pickup(3000) |
| 19:39.00 | elcontrastador | *8 doesn't even show up in the log outside of the beeperr |
| 19:39.12 | *** join/#asterisk danjenkins_ (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 19:39.36 | elcontrastador | i'm thinking this may be some dpma interaction as it does work with the features.conf for parking, etc |
| 19:42.20 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 19:45.41 | *** join/#asterisk BeachBall (~eXcAliBuR@206.162.174.6) |
| 19:47.04 | *** join/#asterisk reconwireless (uid10170@gateway/web/irccloud.com/x-tgmimbpqrcepencb) |
| 19:51.58 | *** join/#asterisk navaismo (~navaismo@201.124.146.234) |
| 19:55.18 | *** join/#asterisk mnathani (~mnathani@umbozero.winvive.com) |
| 19:58.39 | *** join/#asterisk caveat- (hoax@2a01:4f8:191:9111:30::10) |
| 19:58.39 | *** join/#asterisk dtcrshr (~datacrush@unaffiliated/datacrusher) |
| 19:58.39 | *** join/#asterisk SunTsu (miyamoto@unaffiliated/suntsu) |
| 19:58.39 | *** join/#asterisk Frojoe (Frojoe@2a01:7e00::f03c:91ff:fe70:bc74) |
| 19:58.39 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 19:58.39 | *** join/#asterisk Sjors (~sgielen@foo.kassala.de) |
| 19:59.11 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 19:59.38 | *** join/#asterisk ThatCantBe (~irc@dsl.dyn-206.53.180.155.tbinet.bm) |
| 20:00.02 | *** join/#asterisk caveat- (hoax@2a01:4f8:191:9111:30::10) |
| 20:00.57 | *** join/#asterisk serafie1 (~erin@nat/digium/session) |
| 20:01.06 | *** join/#asterisk fling_ (~fling@fsf/member/fling) |
| 20:01.44 | *** join/#asterisk serafie1 (~erin@nat/digium/x-jnvobfrofyxfwxmx) |
| 20:01.52 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 20:02.00 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 20:02.00 | *** join/#asterisk cmendes0101 (~cmendes01@office.phone.com) |
| 20:02.09 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-zbhohsgoyrejmrbr) |
| 20:02.09 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 20:02.34 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
| 20:02.59 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-dfitwvqzqsjfwyqf) |
| 20:03.03 | *** join/#asterisk eirirs (eirirs@dalvik.ping.uio.no) |
| 20:04.49 | *** join/#asterisk dwayne (~dwayne@c-71-207-208-112.hsd1.al.comcast.net) |
| 20:05.26 | *** join/#asterisk dtcrshr (~datacrush@unaffiliated/datacrusher) |
| 20:05.26 | *** join/#asterisk SunTsu (miyamoto@unaffiliated/suntsu) |
| 20:05.26 | *** join/#asterisk Frojoe (Frojoe@2a01:7e00::f03c:91ff:fe70:bc74) |
| 20:05.26 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 20:05.26 | *** join/#asterisk Sjors (~sgielen@foo.kassala.de) |
| 20:06.09 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 20:12.56 | *** join/#asterisk lanning (~lanning@50-193-22-25-static.hfc.comcastbusiness.net) |
| 20:13.56 | *** join/#asterisk Sjors (~sgielen@foo.kassala.de) |
| 20:14.06 | *** join/#asterisk Milarepa (~Milarepa@host-74-211-92-125.beyondbb.com) |
| 20:19.51 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/session) |
| 20:20.11 | *** join/#asterisk tzafrir_laptop (~tzafrir@81.218.177.19) |
| 20:21.04 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-azefwsfakgrixfbx) |
| 20:22.56 | *** join/#asterisk elcontrastador (~textual@12.226.100.130) |
| 20:23.59 | *** join/#asterisk julgr (~julgr@38.104.125.2) |
| 20:40.32 | elcontrastador | penguin: can you explain this? http://pastebin.com/krpJhiT6 |
| 20:41.33 | elcontrastador | I can pickup extension 3000 which dials 3910&3920. Can't pickup 3910 from 3920... |
| 20:46.40 | *** join/#asterisk kayatwork (~kayfox@orca.zerda.net) |
| 20:52.59 | *** join/#asterisk bkruse (~Adium@24.42.229.8) |
| 20:57.54 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 21:10.09 | *** join/#asterisk g_r_eek (~g_r_eek@178-241-228.dynamic.cyta.gr) |
| 21:19.17 | *** join/#asterisk cmendes0101| (~cmendes01@office.phone.com) |
| 21:26.05 | *** join/#asterisk kruftindustries (4016eaec@gateway/web/freenode/ip.64.22.234.236) |
| 21:26.59 | kruftindustries | Would anyone happen to know how to keep the calling line open after the dialed party hangs up with Dial()? |
| 21:27.17 | WIMPy | core show application dial |
| 21:27.44 | kruftindustries | <PROTECTED> |
| 21:27.54 | Qwell | blinks |
| 21:27.58 | *** join/#asterisk jpoz_ (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 21:27.59 | MaliutaLap | don't call hangup() straight after dial() |
| 21:28.03 | kruftindustries | nvm |
| 21:28.12 | kruftindustries | I don't have it in there, I have a goto |
| 21:28.20 | WIMPy | MaliutaLap: That doesn't work. |
| 21:28.39 | WIMPy | You have to tell Dial to continue. |
| 21:28.48 | MaliutaLap | WIMPy: if the next thing after dial() was - let's say - playback() ... |
| 21:28.53 | WIMPy | By default it ends the calling channel as well. |
| 21:29.10 | MaliutaLap | the line would stay open - just not connected to the dialed party |
| 21:29.24 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 21:29.59 | kruftindustries | I have same => n,Dial(${number}) same => n,Goto(101-campaign,1,1) |
| 21:30.24 | kruftindustries | It closes the channel when the called party hangs up |
| 21:30.48 | WIMPy | That's what I said. |
| 21:31.04 | kruftindustries | I didn't see your reply, sorry |
| 21:31.30 | kruftindustries | So, I'm assuming there's a flag I have to set |
| 21:31.44 | kruftindustries | I'm looking through core show application dial |
| 21:31.48 | WIMPy | That's also something I just said. |
| 21:31.56 | WIMPy | And again. |
| 21:33.30 | *** join/#asterisk julgr (~julgr@38.104.125.2) |
| 21:33.57 | kruftindustries | I know you did |
| 21:34.06 | kruftindustries | I only said I was reading it looking |
| 21:34.43 | kruftindustries | It only has options for doing things with the called party after the caller hangs up, not the other way around |
| 21:35.09 | WIMPy | That's ok. Others take half an hor or even a full one to find out what I told them immediately. |
| 21:35.12 | *** join/#asterisk tuxd00d (~tuxd00d@ip24-251-34-116.ph.ph.cox.net) |
| 21:35.38 | WIMPy | You must have missed it. |
| 21:35.44 | navaismo | like: wimpy i cant have queue_log in a file and in the db :( |
| 21:36.17 | kruftindustries | Ok the only thing I see that is remotely close to what I need is the macro CONTINUE |
| 21:36.49 | navaismo | are you using option M? I guess you need option g or was G |
| 21:37.30 | WIMPy | g |
| 21:37.45 | WIMPy | Nothing more. Nothing less. |
| 21:37.52 | *** join/#asterisk lanning (~lanning@50-193-22-25-static.hfc.comcastbusiness.net) |
| 21:39.04 | kruftindustries | Dear gosh |
| 21:39.08 | kruftindustries | It's all so simple |
| 21:39.12 | kruftindustries | :D |
| 21:39.23 | kruftindustries | How did I miss that lol |
| 21:39.29 | WIMPy | coughs |
| 21:39.30 | kruftindustries | Or WIMPy's replies |
| 21:39.36 | navaismo | all reduces to READ |
| 21:39.39 | kruftindustries | Yes yes, |
| 21:39.49 | navaismo | happens a lot |
| 21:40.10 | kruftindustries | I've never had to do that before |
| 21:40.10 | WIMPy | Unfortunatly most Asterisk things aren't remotely as easy as that one. |
| 21:40.14 | kruftindustries | Thanks for pointers lol |
| 21:40.27 | WIMPy | You never had to read? *eg* |
| 21:40.42 | kruftindustries | Yeah it took me a while to figure out how to use values form the DBPut |
| 21:41.01 | WIMPy | Still simple. |
| 21:41.13 | kruftindustries | Yeah you have to use set |
| 21:41.25 | kruftindustries | set(dbget |
| 21:41.56 | kruftindustries | At least agi is easy |
| 21:41.57 | WIMPy | But I have no freaking idea what the funny things dahdi is doing to me are meant to tell me :-( |
| 21:42.45 | kruftindustries | so, why did they take out chan_usbradio and other ham radio related stuff? |
| 21:43.11 | WIMPy | Probably because noone kept them compatible to newer versions. |
| 21:43.19 | kruftindustries | :/ |
| 21:44.01 | WIMPy | Just like chan_capi is (or was) unavailable for current versions. |
| 21:44.15 | *** join/#asterisk dwayne (~dwayne@c-71-207-208-112.hsd1.al.comcast.net) |
| 21:44.45 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 21:45.08 | elcontrastador | I am struggling like heck with simple call pickups. Can someone please take a look at this? http://pastebin.com/3dtZqPw9 |
| 21:45.31 | *** join/#asterisk gusto (~gusto@2a02:810d:8600:8d4:21b:63ff:fe31:8426) |
| 21:45.56 | gusto | so |
| 21:46.04 | gusto | Penguin: hi |
| 21:46.06 | gusto | WIMPy: hi |
| 21:46.17 | gusto | WIMPy: are you making progress with your ISDN card? |
| 21:46.28 | WIMPy | No :-( |
| 21:46.35 | gusto | lol |
| 21:46.41 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 21:49.01 | WIMPy | And I have orders for two more boxes and no good source for cards :-( |
| 22:00.40 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 22:05.26 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:15.01 | *** join/#asterisk rekrul (62e818d8@gateway/web/freenode/ip.98.232.24.216) |
| 22:16.20 | rekrul | So I'm having a bit of trouble connecting to google voice, I keep getting the following error message: Unable to add Google ICE candidates as ICE support not available or no candidates available. I have enabled icesupport in rtp.conf |
| 22:16.30 | rekrul | I don't understand what the issue is |
| 22:17.02 | paulc | not sure about ICE references.. but isn't Google Voice going away in a few months? |
| 22:17.31 | WIMPy | That's what they say. |
| 22:20.11 | rekrul | what? |
| 22:22.04 | rekrul | Who is saying google voice is going away? |
| 22:26.15 | file | http://blog.obihai.com/2013/10/important-message-about-google-voice.html and chan_motif uses the XMPP support |
| 22:36.40 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 22:39.16 | *** join/#asterisk fornax (~fornax@85.183.53.64) |
| 22:41.26 | smkelly | file file file file |
| 22:41.46 | *** kick/#asterisk [smkelly!~file@asterisk/developer-and-muffin-lover/file] by file (smkelly) |
| 22:41.56 | *** join/#asterisk smkelly (~smkelly@mykonos.smkelly.org) |
| 22:42.18 | file | smkelly, hi hi hi |
| 22:42.22 | file | smkelly, tacos |
| 22:42.55 | smkelly | I need an adult! |
| 22:43.08 | file | smkelly, well then don't go to vish |
| 22:43.15 | smkelly | watches file show up on an episode of To Catch a Canadian |
| 22:43.47 | *** join/#asterisk fornax (~fornax@85.183.53.64) |
| 22:45.44 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:48.14 | *** join/#asterisk fornax (~fornax@85.183.53.64) |
| 22:53.23 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 22:54.32 | fornax | hi, still someone here who can give me a hint concerning a dahdi sound problem? |
| 22:58.01 | Kobaz | mm |
| 22:58.04 | Kobaz | i gots a drive dieing |
| 23:02.06 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 23:26.00 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 23:29.39 | *** join/#asterisk Ta^3 (~tacvbo@187.188.107.19) |
| 23:34.48 | *** join/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002) |
| 23:34.59 | mattwj2002 | hi all |
| 23:35.00 | mattwj2002 | :) |
| 23:42.20 | *** join/#asterisk Bonte (~oryx@bark.ungulate.net) |
| 23:43.05 | Bonte | Has anyone ever gotten the chat/messaging function of a SIP client to work through Asterisk? Namely Zoiper (Android app) |
| 23:43.28 | *** join/#asterisk bkruse (~Adium@74.51.115.113) |
| 23:45.24 | mattwj2002 | hi Bonte |
| 23:46.21 | [TK]D-Fender | Bonte: * is capable of processing SIP MESSAGE events. |
| 23:47.38 | SpeedEvil | Do SIP providers typically gateway SMSs to SIP message? |
| 23:47.48 | Bonte | [TK]D-Fender, yeah I thought so.. I think Zoiper isn't following the specs correctly |
| 23:48.07 | [TK]D-Fender | Bonte: got backup for that? |
| 23:48.09 | Bonte | Every time a message hits *, it complains |
| 23:48.16 | [TK]D-Fender | Bonte: Show us |
| 23:48.19 | Bonte | k! |
| 23:48.20 | [TK]D-Fender | ~pb |
| 23:48.21 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 23:48.23 | [TK]D-Fender | ^^^^^ |
| 23:48.28 | Bonte | Yup |
| 23:50.58 | Bonte | http://pastebin.com/5SLrmg4p |
| 23:51.24 | Bonte | Error on line 21 & 22 |
| 23:53.09 | [TK]D-Fender | [Feb 26 18:49:24] WARNING[22895][C-00000000]: app_dial.c:2435 dial_exec_full: Unable to create channel of type 'SIP' (cause 58 - Bearer capability not available) |
| 23:53.26 | [TK]D-Fender | app_dial is being called... this means you are processing DIALPLAN... and clearly not looking at the verbose of it |
| 23:53.39 | [TK]D-Fender | You are processing this like a voice call.. |
| 23:53.46 | [TK]D-Fender | cor set verbose 10 |
| 23:53.47 | [TK]D-Fender | ^ |
| 23:53.48 | Bonte | Ah, that won't work |
| 23:53.49 | [TK]D-Fender | core* |