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| 00:32.55 | Thedr | Is it possible to route incoming calls based on thier caller ID? |
| 00:33.04 | leifmadsen | yes |
| 00:34.46 | Thedr | is it just a matter of putting thhe number into the caller ID match field? |
| 00:35.17 | leifmadsen | you can either use ${CALLERID(num)} to match in the dialplan, or you can do something like: |
| 00:35.38 | leifmadsen | exten => 4165551212/9052340000,1,NoOp() |
| 00:35.48 | Thedr | thanks |
| 00:35.48 | leifmadsen | after the / that is the CID match |
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| 01:27.04 | pushpop | Hello, does anyone use flowroute here? |
| 01:27.40 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 01:27.47 | WIMPy | ~ask |
| 01:27.47 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 01:28.15 | pushpop | Hello, does anyone use flowroute here? I was wondering how you setup the caller id you with to Transmit when you purchase a DID from them. |
| 01:28.16 | carrar | I use Asterisk! |
| 01:28.29 | pushpop | wish* |
| 01:29.14 | carrar | Set(CALLERID(number)=8675309) ?? |
| 01:31.37 | carrar | s/number/num/ |
| 01:31.47 | pushpop | ok |
| 01:31.48 | pushpop | thx |
| 01:31.59 | pushpop | but what about a company name |
| 01:32.02 | pushpop | like Company A |
| 01:32.28 | carrar | You need to have them set that up |
| 01:32.36 | pushpop | flowroute will do that? |
| 01:32.46 | carrar | You can send it, but it will only be good for people probably connected to them via SIP |
| 01:32.55 | carrar | call them |
| 01:33.06 | carrar | Tell them what you want your Caller ID Name to be |
| 01:33.08 | pushpop | yea thats no good |
| 01:33.16 | carrar | they will need to set that up for you |
| 01:33.18 | pushpop | need everyone to see the company |
| 01:33.21 | pushpop | OK |
| 01:34.27 | pushpop | you know of any sip providers in the US that this can be done for sure? |
| 01:34.49 | carrar | What is "this" ? |
| 01:35.07 | pushpop | transmit caller id with name |
| 01:35.11 | carrar | Allow you to send whatever you want for a caller ID Name to show up on remote peoples phones? |
| 01:35.25 | pushpop | yea or choose a name when you purchase the DID |
| 01:35.31 | carrar | As long as all these people are on YOUR PBX you can do that |
| 01:35.50 | pushpop | well my customers would not be on my pbx |
| 01:35.56 | carrar | telco's make a DIP into a database to get that caller id name |
| 01:36.09 | pushpop | yea |
| 01:36.22 | pushpop | is there a sip provider that does that by default? |
| 01:37.02 | carrar | How would that work? |
| 01:37.13 | WiretapWork_ | pushpop, most telcos don't transmid CIDNAME between each other |
| 01:37.27 | *** join/#asterisk Kumbang (~kumbang@180.245.137.5) |
| 01:37.59 | pushpop | say if I have a verizon number for my home line. Everyone I call no matter their carrier sees my name on their caller id |
| 01:38.18 | pushpop | this is not possible with sip providers? |
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| 01:49.18 | *** join/#asterisk lwizardl (~james@c-68-60-84-225.hsd1.mi.comcast.net) |
| 01:49.20 | WiretapWork_ | pushpop, call your provider and find out |
| 01:49.21 | lwizardl | hello |
| 01:50.31 | lwizardl | is there anyway to use a regular phone modem for a pbx? |
| 01:50.55 | lwizardl | or do I fully need a fxo card for a single line |
| 01:51.21 | WIMPy | That's not supported any more. |
| 01:51.31 | lwizardl | WIMPy, what aint? |
| 01:51.50 | lwizardl | the modem or the fxo |
| 01:52.02 | *** join/#asterisk coppice (~chatzilla@m121-202-97-173.smartone-vodafone.com) |
| 01:52.11 | WIMPy | modems |
| 01:52.23 | lwizardl | thats lame |
| 01:52.30 | WiretapWork_ | not really |
| 01:52.34 | WiretapWork_ | modems aren't designed for voice |
| 01:52.49 | WiretapWork_ | good at making beeps and boops but they do a piss poor job of representing the human voice |
| 01:52.59 | WIMPy | Most of them were. |
| 01:54.01 | lwizardl | all i know is the 3com and usrobotics ones i used to use i kept a phone line handset attached to it 24hrs a day and I never noticed any issues |
| 01:54.16 | *** join/#asterisk mzb (~mzb@120.23.172.162) |
| 01:55.24 | WIMPy | It used to work very well for me as well, but that was aaaaaages ago. |
| 01:55.42 | lwizardl | WIMPy, same this was like 1996-1998ish |
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| 01:55.46 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 01:56.54 | WIMPy | At that time als the analog stuff had been replaced by ISDN. |
| 01:57.16 | WIMPy | s/als/all/ |
| 01:58.14 | lwizardl | yeah i know. |
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| 02:06.34 | lwizardl | what would be the cheapest cards that work with asterisk now, the linksys pap2 ? |
| 02:08.03 | WiretapWork_ | lwizardl, the PAP2 and PAP2T only provide FXS |
| 02:08.26 | WiretapWork_ | not FXO |
| 02:08.31 | WiretapWork_ | for that you need an SPA3000 or 3102 |
| 02:08.49 | lwizardl | whats the difference between the two fxo/fxs ? |
| 02:10.11 | WiretapWork_ | FXO = Foreign Exchange Office, is what your telephone is, essentially, or your computer's modem |
| 02:10.49 | WiretapWork_ | FXS = Foreign Exchange Station, functions like the line port at the exchange, provides the power to the phone and supplies ringing voltages, etc |
| 02:12.17 | lwizardl | so for using a voip only system which would be needed fxo or fxs ? |
| 02:13.39 | *** join/#asterisk nny (~Scott_2@174.107.201.103) |
| 02:13.51 | nny | anyone know off the top of their head which XML line in the polycomm config sets timezone? |
| 02:14.49 | nny | nm lol |
| 02:14.54 | nny | my google was weak at first [root@asterisk01 tftpboot]# grep "tcpIpApp.sntp.gmtOffset" * |
| 02:14.54 | nny | sip.cfg: tcpIpApp.sntp.gmtOffset="-32400" |
| 02:14.54 | nny | sip.cfg: tcpIpApp.sntp.gmtOffset.overrideDHCP="1" |
| 02:14.54 | nny | sip.remote.cfg: tcpIpApp.sntp.gmtOffset="-32400" |
| 02:14.54 | nny | sip.remote.cfg: tcpIpApp.sntp.gmtOffset.overrideDHCP="1" |
| 02:14.56 | nny | oops! |
| 02:14.58 | nny | crap |
| 02:14.59 | nny | sorry |
| 02:15.02 | nny | leaving >< |
| 02:15.03 | pigpen | sip.cfg |
| 02:15.16 | nny | meant to only paste one line dammit |
| 02:15.23 | nny | apologies again for spamming |
| 02:15.24 | pigpen | ^^^ what he said...just much, much more...heh |
| 02:15.25 | nny | ty pigpen |
| 02:15.39 | pigpen | hey, I've done it. |
| 02:16.27 | pigpen | nny, fyi, you can set it to honor the dhcp settings, but just set the ntp server info in the cfg file, dhcp can be a bit hit and miss. |
| 02:20.30 | WiretapWork_ | lwizardl, for a voip only system, neither |
| 02:20.41 | WiretapWork_ | if you're getting your calls coming in over a SIP trunk |
| 02:20.47 | WiretapWork_ | (or IAX2) |
| 02:23.34 | lwizardl | yeah |
| 02:28.12 | lwizardl | i think they are just sip accounts |
| 02:29.04 | WiretapWork_ | then why do you want a 'card' for it? |
| 02:29.17 | WiretapWork_ | voip by its nature runs over IP |
| 02:29.37 | lwizardl | i thought you had to use the cards for conecting a handset to the system. |
| 02:30.00 | WiretapWork_ | .... then it wouldn't be a 'voip only system' |
| 02:30.03 | WiretapWork_ | if you're not using voip phones |
| 02:30.08 | WiretapWork_ | then there's a PSTN element |
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| 03:02.24 | *** join/#asterisk kaushal (~kaushal@49.248.16.122) |
| 03:02.26 | kaushal | Hi |
| 03:02.28 | carrar | <PROTECTED> |
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| 03:02.48 | WIMPy | lo |
| 03:05.31 | kaushal | I have connected Digium 2 span PRI Card to the system |
| 03:05.46 | kaushal | when i run zap show status at CLI Prompt |
| 03:05.59 | WIMPy | You should upgrade. |
| 03:06.03 | kaushal | No such command 'zap show status' (type 'core show help zap show' for other possible commands) |
| 03:06.05 | carrar | heh |
| 03:06.24 | carrar | need to load the modules |
| 03:06.30 | carrar | and drivers probably |
| 03:06.36 | WIMPy | Probably uourself then :-) zaptel has been renamed dahdi some years ago. |
| 03:07.03 | carrar | http://www.asterisk.org/downloads |
| 03:07.09 | kaushal | http://pastebin.ubuntu.com/609300/ |
| 03:07.25 | carrar | 1.8.4 Asterisk Communications Engine has your name all over it! |
| 03:07.31 | carrar | ALL OVER I |
| 03:07.32 | carrar | T |
| 03:07.42 | kaushal | carrar: not sure i understand that |
| 03:07.48 | carrar | graffiti style |
| 03:08.20 | kaushal | carrar: Please guide |
| 03:08.23 | WIMPy | kaushal: dahdi show status |
| 03:08.27 | kaushal | ok |
| 03:08.32 | carrar | what is zap in your command output? |
| 03:08.33 | WIMPy | see above |
| 03:09.01 | kaushal | No such command 'dahdi show status' (type 'core show help dahdi show' for other possible commands) |
| 03:09.09 | kaushal | -bash: dahdi: command not found |
| 03:09.14 | carrar | Why are you using the word "zap"? |
| 03:09.17 | pigpen | anybody around that has experience with audiocodes? |
| 03:09.28 | WIMPy | Did you configure it? |
| 03:09.37 | pigpen | heh. yeah. |
| 03:09.47 | kaushal | WIMPy: ? |
| 03:09.55 | carrar | try: dahdi show status |
| 03:10.00 | pigpen | I think I am having a glare problem. |
| 03:10.11 | kaushal | carrar: on the CLI prompt ? |
| 03:10.12 | carrar | since you are using dahdi drivers |
| 03:10.16 | WIMPy | Did you edit chan_adahdi.conf to your situation? |
| 03:10.18 | carrar | yes pls k thanks |
| 03:10.22 | kaushal | WIMPy: yes |
| 03:10.30 | pigpen | ie: calls come in channels 1,2,3,4, but for some reason I cannot get the bastard audiocodes to dial out 4,3,2,1 |
| 03:10.47 | pigpen | I set everything to descending that I can, but no dice. |
| 03:10.47 | WIMPy | kaushal: Then look at what 'module locad chan_dahdi' tells you. |
| 03:11.40 | kaushal | WIMPy: do you want output of lsmod ? |
| 03:11.52 | carrar | kaushal, report pls |
| 03:11.57 | WIMPy | No. |
| 03:11.59 | carrar | on the command |
| 03:12.13 | pigpen | kaushal, I think he is saying that the dahdi module may not be loaded. |
| 03:12.15 | *** join/#asterisk g00gle (~thameema@c-98-248-232-219.hsd1.ca.comcast.net) |
| 03:12.20 | carrar | try: dahdi show status |
| 03:12.32 | pigpen | if no "dahdi" command in the cli, the module is not loaded or compiled. |
| 03:12.45 | kaushal | carrar: http://pastebin.ubuntu.com/609302/ |
| 03:12.57 | pigpen | don't say the word "zap" you will likely get shot. |
| 03:13.04 | carrar | Thats not what I asked |
| 03:13.24 | carrar | try: dahdi show status |
| 03:13.45 | WIMPy | kaushal: Are you sure you want all that stuff loaded? |
| 03:14.04 | carrar | You're going there :) |
| 03:14.12 | pigpen | example: this is my output: |
| 03:14.13 | pigpen | Description Alarms IRQ bpviol CRC4 Fra Codi Options LBO |
| 03:14.13 | pigpen | Wildcard TDM400P REV I Board 5 OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) |
| 03:14.18 | kaushal | -bash: dahdi: command not found |
| 03:14.21 | pigpen | ^^^ this is what he is looking for. |
| 03:14.28 | carrar | but will never get |
| 03:14.28 | pigpen | then you don't have dahdi loaded. |
| 03:14.54 | pigpen | kaushal, as he noted above, type in "module load chan_dahdi" |
| 03:14.59 | carrar | cause although I type it in IRC, it never makes it to his screen |
| 03:15.03 | pigpen | this will attempt to load the dahdi module. |
| 03:15.16 | WIMPy | You do 'ahdi show status' a the *CLI (rasterisk). |
| 03:15.36 | pigpen | if this doesn't work, then your system is not likely compiled correctly. |
| 03:15.44 | carrar | changes mode on #Asterisk for kaushal +glasses |
| 03:15.55 | WIMPy | From my typing, I think I get tired. |
| 03:16.05 | kaushal | pigpen: module load chan_dahdi ? |
| 03:16.12 | kaushal | its CentOS |
| 03:16.14 | WIMPy | kaushal: How did you install Asterisk? |
| 03:16.33 | pigpen | kaushal, can you get into the asterisk cli. <<< basics. |
| 03:16.41 | pigpen | Yes - No (circle one) |
| 03:16.41 | carrar | ~thebook |
| 03:16.41 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 03:16.48 | carrar | pls read |
| 03:16.48 | kaushal | http://pastebin.ubuntu.com/609303/ |
| 03:16.49 | carrar | ALL |
| 03:17.00 | kaushal | pigpen: yes |
| 03:17.34 | pigpen | great, in the asterisk cli, type this: module load chan_dahdi.so |
| 03:17.39 | WIMPy | kaushal: Don't post random things. It you want to get on, aswer our questions. Or maybe better go read a bit first. |
| 03:17.55 | kaushal | http://pastebin.ubuntu.com/609305/ |
| 03:18.07 | kaushal | WIMPy: I am following your suggestion |
| 03:18.17 | pigpen | kaushal, now, this is assuming you are using a modern asterisk version. If it is old, then get new, and start over. |
| 03:18.22 | WIMPy | Ok, and now 'module load chan_dahdi'. |
| 03:18.29 | carrar | http://pastebin.ubuntu.com/609306/ |
| 03:18.43 | WIMPy | Argh |
| 03:19.10 | pigpen | I am going to see if my ice pick will fit nicely in my forhead. |
| 03:19.30 | carrar | Now check this out |
| 03:19.31 | carrar | http://pastebin.ubuntu.com/609310/ |
| 03:19.33 | carrar | pretty cool |
| 03:19.38 | WIMPy | kaushal: and then there was the "How did you install Asterisk?" bit. |
| 03:19.39 | carrar | LOOPED |
| 03:19.47 | pigpen | haha....funny. |
| 03:19.53 | kaushal | WIMPy: sure |
| 03:19.55 | carrar | heh |
| 03:19.55 | *** join/#asterisk Kumbang (~kumbang@180.245.137.5) |
| 03:21.03 | pigpen | http://pastebin.ubuntu.com/609314/ |
| 03:21.13 | pigpen | ^^^ this is exactly what you need. |
| 03:21.17 | pigpen | ;-) |
| 03:21.18 | carrar | haha |
| 03:21.28 | carrar | That link has a virus init! |
| 03:21.35 | carrar | trixbox virus! |
| 03:21.41 | pigpen | yeah, I thought you would get a kick out of it. |
| 03:21.43 | WIMPy | Brainfuck |
| 03:21.58 | kaushal | WIMPy: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-YUM%28CentOS%2FRedHat%29 |
| 03:22.04 | pigpen | You know, I loaded it once, just for the hell of it. |
| 03:22.17 | pigpen | yumm...... |
| 03:22.26 | pigpen | yum is for sissys |
| 03:22.39 | pigpen | man up and use portage on gentoo. |
| 03:22.56 | WIMPy | But I'd assume it contains chan_dahdi. So it's probably an error in chan_dahdi.conf. |
| 03:23.00 | carrar | compile from source! |
| 03:23.14 | kaushal | http://pastebin.ubuntu.com/609315/ |
| 03:23.16 | pigpen | you haven't lived until you have beaten your head bloody trying to get some senseless package to compile.... |
| 03:23.50 | pigpen | kaushal, you need to configure dahdi. |
| 03:24.01 | pigpen | ~thebook |
| 03:24.02 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 03:24.15 | carrar | probably need to config it |
| 03:24.51 | kaushal | pigpen: please be calm |
| 03:24.56 | kaushal | dont abuse |
| 03:25.09 | pigpen | your are serious. |
| 03:25.17 | pigpen | ^^^that was a question. |
| 03:25.34 | carrar | There is even a chapter on dadhi |
| 03:25.40 | kaushal | carrar: ok |
| 03:25.42 | kaushal | will read it |
| 03:25.55 | kaushal | carrar: is there a PDF version of it ? |
| 03:25.59 | carrar | yes |
| 03:26.16 | carrar | WHEN you purchase it you can download a PDF and several other version of it |
| 03:26.27 | pigpen | http://pastebin.ubuntu.com/609318/ |
| 03:26.34 | pigpen | ^^^ for the pastebin lovers out there. |
| 03:26.38 | pigpen | later. |
| 03:27.10 | carrar | Please ring the bell when you purchase it |
| 03:28.14 | carrar | http://oreilly.com/catalog/9780596517342/ |
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| 04:21.31 | dan__t | Alright, still not able to get this working. I have MixMonitor running, it records a call just fine, places it exactly where I wnat it to. However, when I try to run <command> as an argument to MixMonitor, the script never gets ran. Here's what I'm dealing with: http://pastebin.com/XYVFaZL8 I can run the command with the argument as the asterisk user on the command line and it works perfectly fine. |
| 04:21.44 | *** part/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 04:22.13 | dan__t | I was thinking it had something to do with asterisk running chroot'ed, since this is Digium's RHEL package, but if that were the case, the call wouldn't end up under /var/recordedcalls/inprogress. I don't get it. |
| 04:28.21 | kaldemar | dan__t: can you run it from CLI with !command? |
| 04:29.52 | dan__t | I don't understand, sorry? |
| 04:30.30 | kaldemar | !/path/to/script args |
| 04:35.50 | dan__t | well yeah, it works |
| 04:36.40 | dan__t | What specifically am I looking for? |
| 04:37.20 | dan__t | hmmmm |
| 04:37.29 | dan__t | This may just be me being stupid. One sec. |
| 04:45.36 | dan__t | Nope, not me this time. |
| 04:45.42 | dan__t | Not sure what the deal is. |
| 04:45.49 | dan__t | Though I am new to this. |
| 04:48.22 | *** join/#asterisk fauxalliance (~fauxallia@142.163.147.26) |
| 04:50.53 | dan__t | script exits 0 as it should, everything goes fine, everything works |
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| 04:56.19 | *** join/#asterisk benngard (~mabe@213.88.138.230) |
| 05:11.20 | dan__t | I just don't get it. |
| 05:13.40 | kaldemar | is your script lacking paths to commands? |
| 05:14.26 | dan__t | It is not. |
| 05:15.23 | dan__t | I've tested with /bin/touch /tmp/test as my first line after the shebang, to no avail. |
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| 05:52.05 | dan__t | I also tried System() as command, thinking that was it... no dice. |
| 05:59.33 | dan__t | meh. got it ot work. |
| 05:59.50 | dan__t | I had <command> enclosed in quotes because I was using ${UNIQUEID}, just figured it was the right thing to do. |
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| 06:11.02 | dimm | how to know adress of sip provider connected with asterisk ? |
| 06:11.36 | dan__t | show sip peers? |
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| 06:12.34 | kaldemar | dimm: connected how? are they registering to you? |
| 06:16.19 | dimm | kaldemar, hi, yes, they are registering to me. |
| 06:17.23 | dimm | 'sip show peers' showing something... i will analyze |
| 06:17.35 | kaldemar | dimm: if you've defined a peer that matches with host=dynamic and they have registered, sip show peers will show an address. |
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| 06:39.03 | zkn | question about execif |
| 06:39.05 | zkn | ExecIf(expression?appiftrue(args)[:appiffalse(args)]) |
| 06:39.39 | zkn | would it be posible to write true and false on two different lines somehow? |
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| 06:40.51 | zkn | i think according to the book, i should be able to to ExecIf(expression?appiftrue(args) on one line and ExecIf(expression?:appiffalse(args) on the other line |
| 06:41.13 | zkn | for the same expression, I mean |
| 06:41.22 | kaldemar | zkn: sure. just leave the false option out of the first line and the true out of the second. |
| 06:43.24 | kaldemar | you'll get a warning when there is no application to run on true. put a noop in there for example. |
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| 06:52.50 | zkn | kaldemar: yesyes, that's what was confusing me at first |
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| 07:22.18 | manawenuz | Hi everybody , |
| 07:22.36 | manawenuz | can somebody help me with RetryDial() Command in asterisk ? |
| 07:23.27 | WiretapSeven | ~ask |
| 07:23.28 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 07:23.38 | kaldemar | did you read what "core show application RetryDial" says? |
| 07:24.29 | ChannelZ | It says you are to make chocolate chip cookies for us. |
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| 07:26.05 | manawenuz | kaldemar: yes i did , the problem is that , i'm trying to achieve autoredial in case of unavailable free dahdi channel or busy number , in my dialplan , but with no luck so far , i followed core show application retrydial , and no luck as of yet |
| 07:26.35 | manawenuz | kaldemar: here's what i did : ;exten => _9.,n,RetryDial(please-wait,5,5,DAHDI/g0/${EXTEN:1},60,md) |
| 07:28.03 | ChannelZ | is it commented out for a reason? |
| 07:28.04 | manawenuz | kaldemar: when i call a number , it plays MOH , and then it starts to play the busy tone , for 10 second or so , and then channel hangs up , nothing happens. as far as i understood it shouldn't play the busy tone at all , and it should redial 5 times , but it doesn't . any ideas ? |
| 07:28.13 | ChannelZ | And what is it doing? |
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| 07:28.44 | manawenuz | ChannelZ: definitely it is commented on purpose at the moment , i'm not using it ATM , since it doesn't work |
| 07:28.51 | ChannelZ | wait, this is a *remote* busy signal? |
| 07:30.02 | kaldemar | manawenuz: how about using a queue? sounds like that's the behavior you're after. |
| 07:30.06 | manawenuz | ChannelZ: what is a remote busy signal |
| 07:30.29 | manawenuz | kaldemar: i've never used queue , if you can guide me through , it'd be lovely |
| 07:30.31 | ChannelZ | you say you're hearing a busy tone.. is that a result of the number dialed being busy? |
| 07:31.03 | manawenuz | ChannelZ: yes ,it is , actually , i'm calling a number which is on purpose busy |
| 07:31.31 | ChannelZ | and is this a T1 or POTS? |
| 07:31.54 | ChannelZ | Because if it's POTS then asterisk has no idea |
| 07:32.23 | ChannelZ | unless you use busydetect which is likely to cause more problems than it solves |
| 07:32.25 | kaldemar | manawenuz: http://ofps.oreilly.com/titles/9780596517342/asterisk-ACD.html |
| 07:33.55 | manawenuz | ChannelZ: it's an FXO |
| 07:34.19 | manawenuz | ChannelZ: and as a matter of fact , it's only a single FXO |
| 07:34.26 | ChannelZ | Well then without turning on busy detection for those channels you really get no call progress |
| 07:35.27 | ChannelZ | As soon as it's done dialing, the channels bridge and it's considered a successful call |
| 07:35.28 | manawenuz | ChannelZ: should i turn on busydetection on dahdi ? |
| 07:37.05 | ChannelZ | You can but it basically is just listening to the entire call for the busy tone.. it's not unheard of for someone's voice to trigger it accidentally in the middle of a call and hang up |
| 07:37.42 | manawenuz | ChannelZ: :)) |
| 07:37.51 | manawenuz | ChannelZ: is there any other way to implement it ? |
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| 07:38.59 | manawenuz | ChannelZ: in system conf ? |
| 07:39.21 | ChannelZ | Besides getting a T1 or some other connection that supports full call progress, using SIP with an ITSP that does instead of a analog phone line |
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| 07:40.38 | ChannelZ | and no it's in chan_dahdi.conf |
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| 07:41.13 | manawenuz | ChannelZ: in chan_dahdi , i can't find anything about hangup detection |
| 07:41.19 | manawenuz | ChannelZ: what is it called ? |
| 07:41.22 | ChannelZ | busydetect |
| 07:41.39 | ChannelZ | and there's a couple other options that start with busy |
| 07:41.59 | ChannelZ | brb |
| 07:42.02 | manawenuz | busydetect is already on |
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| 08:00.23 | ectospasm | manawenuz: make sure loadzone and defaultzone are set correctly in /etc/dahdi/system.conf, make sure opermode is set correctly for the particular dahdi driver you're using, and set busydetect=yes, busycount=6 in /etc/asterisk/chan_dahdi.conf |
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| 08:01.40 | manawenuz | ectospasm: localzone and defaultzone are us |
| 08:01.54 | manawenuz | ectospasm: busy detect in chan_dahdi.conf is also configured |
| 08:02.04 | manawenuz | ectospasm: what is upermode ? |
| 08:02.25 | ectospasm | manawenuz: opermode tells the driver what country you're in. |
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| 08:03.08 | ectospasm | manawenuz: edit /etc/modprobe.d/dahdi.conf, make sure "options wctdm24xxp opermode=US" and restart dahdi |
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| 08:03.46 | ectospasm | manawenuz: note, if you're not in the US you should change it to your location, or else this stuff likely isn't ever gonna work for you. |
| 08:04.23 | manawenuz | ectospasm: this server is situated in iran , and when i put ir instead of us |
| 08:04.28 | manawenuz | ectospasm: it didn't work at all |
| 08:04.36 | ectospasm | manawenuz: yeah, there you go |
| 08:04.49 | manawenuz | ectospasm: [root@localhost asterisk]# /etc/init.d/dahdi restart |
| 08:04.49 | manawenuz | Unloading DAHDI hardware modules: ERROR: Module wctdm is in use |
| 08:04.49 | manawenuz | ERROR: Module dahdi_echocan_mg2 is in use |
| 08:04.49 | manawenuz | ERROR: Module dahdi is in use by dahdi_echocan_mg2,wctdm |
| 08:04.49 | manawenuz | error |
| 08:04.49 | manawenuz | Loading DAHDI hardware modules: |
| 08:04.50 | manawenuz | WARNING: /etc/modprobe.d/dahdi.conf line 5: ignoring bad line starting with 'opermode=US' |
| 08:04.50 | manawenuz | <PROTECTED> |
| 08:04.58 | ectospasm | You'll need to code that in the driver, check out zonedata.c |
| 08:05.07 | ectospasm | manawenuz: don't flood |
| 08:05.20 | ectospasm | manawenuz: I refuse to even consider your questions now. |
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| 08:06.12 | manawenuz | ectospasm: i'm sorry , i'm new to all irc thing , |
| 08:06.18 | ectospasm | manawenuz: you'll have to look at the ITU spec (look in the source file I mentioned) |
| 08:06.18 | manawenuz | ectospasm: didn't meed to flood |
| 08:06.29 | manawenuz | ectospasm: thank you so much |
| 08:06.39 | ectospasm | Good luck! |
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| 08:31.50 | X-Rob | ectospasm, dude, he fucked up the insert. 'line starting with opermode=US' |
| 08:31.52 | X-Rob | _starting with_ |
| 08:31.58 | X-Rob | note: people are idiots. |
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| 08:51.18 | ectospasm | X-Rob: yes, it looks like he couldn't read |
| 08:54.59 | ectospasm | X-Rob: but then again, I didn't even read it. Person needs to learn not to flood. |
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| 09:52.23 | schmidts | hello |
| 09:58.01 | kleszcz | hi |
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| 10:29.28 | WiretapSeven | can asterisk pass an incoming sip call to an internal extension without answering it? |
| 10:31.31 | kaldemar | don't use app Answer in the dialplan before Dial. |
| 10:31.37 | WiretapSeven | sweet |
| 10:32.21 | WiretapSeven | friend of mine is coming onboard the queue for my new project and wants to pipe it through his own PBX |
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| 11:11.00 | Chainsaw | This looks genuine: https://issues.asterisk.org/view.php?id=18898 |
| 11:11.18 | Chainsaw | I'm getting this with an Ekiga 3.2.7 client which is sending out PUBLISH. |
| 11:14.57 | Chainsaw | Anyone happen to know to turn that off? |
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| 11:42.44 | moodyy | has anyone configured asterisk to perform law interception |
| 11:43.06 | leifmadsen | law interception? |
| 11:43.25 | leifmadsen | places an Asterisk box in the middle of the road to interfere with the police chase |
| 11:43.38 | moodyy | of calls |
| 11:43.50 | moodyy | CALEA |
| 11:43.58 | leifmadsen | just use ChanSpy() |
| 11:44.07 | leifmadsen | that's all I did... |
| 11:44.25 | Tim_Toady | 'lawful interception' |
| 11:45.30 | moodyy | in chanSPY i have to make some sort of action ... dial a number and spy a channel, right? |
| 11:45.41 | moodyy | yes 'lawful interception', sorry |
| 11:46.05 | Tim_Toady | thers also monitor() and mixmonitor() that record calls |
| 11:46.05 | moodyy | in my case i want to make this more or less automatic |
| 11:46.20 | leifmadsen | just sounds like dialplan creation then |
| 11:46.36 | leifmadsen | but chanspy() is the app for listening to calls |
| 11:46.50 | leifmadsen | ~asteriskcookbook |
| 11:46.50 | infobot | The Asterisk Cookbook is available for purchase as an eBook from O'Reilly at http://oreilly.com/catalog/0636920018551 or via Amazon. The book is available under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and can be read online at http://ofps.oreilly.com/titles/9781449303822/ |
| 11:46.56 | leifmadsen | probably some useful stuff there |
| 11:47.37 | schmidts | leifmadsen you should add something about connectedline stuff into the cookbook ;) |
| 11:47.46 | leifmadsen | schmidts: I should do a lot of things ;) |
| 11:47.50 | Chainsaw | is a big fan of connected line updates |
| 11:48.00 | leifmadsen | schmidts: A:TDG already has connected line update stuff I'm pretty sure |
| 11:48.11 | Chainsaw | We do CRM lookups on outbound numbers, so now the "Called number" list on the Polycom handsets has names & numbers. |
| 11:48.48 | schmidts | leifmadsen i didnt found it in there but i dont take much time about searching in THE BOOK ;) |
| 11:49.43 | schmidts | Chainsaw i also love this feature but i didnt get it work, they way i understand it should work ;) |
| 11:50.58 | moodyy | yes i know, but i want to send the audio stream to another destination (in the justice system), and i want this to happen to some numbers only |
| 11:52.37 | leifmadsen | moodyy: yes, it is possible, but will require some creative dialplan and manipulation. You may need to use something like Originate(), or an AMI interface to control the setup of the calls and connecting the calls together to listen to certain channels |
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| 11:53.35 | QuantumSchema | Good mornin' all! |
| 11:57.24 | moodyy | hum, ok, i'm going to test some of those applications, thanks |
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| 12:00.42 | QuantumSchema | I'm facing an interesting dilema.... |
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| 12:01.52 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 12:02.30 | QuantumSchema | I've got my AddQueueMember as such: AddQueueMember(6226,Local/${AGENT}@agents/n,0,,${AGENTNAME},Local/${AGENT}@agents/n} but in the queue log it shows "Local/1242@agents/n' for action when a "REMOVEMEMBER" or "ADDMEMBER" is logged. |
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| 12:03.00 | QuantumSchema | If you pause or unpause, however, it shows the agent's name and then "PAUSED" or "UNPAUSED".... |
| 12:03.42 | QuantumSchema | Is there something in the AddQueueMember that needs to be changed so that the ADDMEMBER or REMOVEMEMBER reflects the agent's name? I've added a QueueLog to rectifiy this for the time being but I'm not sure if this is by design. |
| 12:04.04 | leifmadsen | you'd have to look at the code |
| 12:04.14 | leifmadsen | could be an implementation detail that isn't consistent |
| 12:06.22 | QuantumSchema | Hmmmm.... digging through now. |
| 12:06.41 | QuantumSchema | Any thoughts as to which file/dir may contain the source for that bit? |
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| 12:08.17 | QuantumSchema | Think I got it... |
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| 12:11.00 | QuantumSchema | Yup! that's it! |
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| 12:17.41 | QuantumSchema | All lines that use ast_queue_log and reference anything other than REMOVEMEMBER or ADDMEMBER call "interface" for the variable, all other statuses are "membername"..... |
| 12:18.22 | QuantumSchema | Woopps....backwards.... anything other than REMOVEMEMBER or ADDMEMBER = "membername", those with REMOVEMEMBER or ADDMEMBER call "interface"..... |
| 12:18.29 | leifmadsen | what version of asterisk? |
| 12:18.40 | leifmadsen | I vaguely recall that being an issue someone may have reported |
| 12:18.46 | QuantumSchema | 1.8.3... |
| 12:19.10 | leifmadsen | double check 1.8 to see if it's already been resolved (and/or trunk) |
| 12:19.29 | leifmadsen | not sure why, but I seem to remember something like that being fix |
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| 12:22.44 | QuantumSchema | hmmm..... https://issues.asterisk.org/view.php?id=15829 |
| 12:23.01 | QuantumSchema | Hey! You're name is in there leif! LoL |
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| 12:24.09 | leifmadsen | QuantumSchema: only an imposter |
| 12:24.12 | leifmadsen | is the real leifmadsen |
| 12:24.25 | QuantumSchema | Ohh wow! No kiddin? |
| 12:24.49 | JPVLN | Hello, I have a problem with Digium 4 port ISDN BRI card, can somebody help me to solve the problem? |
| 12:24.56 | leifmadsen | wow that patch is OLD! |
| 12:25.15 | leifmadsen | QuantumSchema: I will try and build an updated patch in a bit -- I'm fixing errata |
| 12:26.26 | QuantumSchema | Wow thanks! No worries on speed. I've got a work around at the moment (logging ADD and REMOVE twice, once done manually with the member name). |
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| 13:39.34 | afink | I've been googling for a solution to record all calls made in and out, haven't been able to find much anyone have a good solution? |
| 13:40.00 | leifmadsen | run all calls in and out of the system through a subroutine or macro that enables call recording |
| 13:40.04 | leifmadsen | it's as simple as that |
| 13:40.10 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 13:40.15 | leifmadsen | there is no "recordallcalls=yes" option |
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| 13:41.04 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 13:41.30 | Dovid | anyone know why in Asterisk I set the session sip timer to 90 seconds and it sends the re-invite in 45. i set it to 120 it sends @ 60. always seems to be half of what i set |
| 13:43.42 | leifmadsen | do you actually mean sip session timer, registration, or re-invite? |
| 13:44.06 | Dovid | leifmadsen: Yes I do. |
| 13:44.37 | Dovid | when it does the re-invite (because of the sessionsip timer) in the sip trace you see 90 but the re-invite goes out at 45 |
| 13:44.45 | Dovid | if i change to 12- it goes out @ 60 |
| 13:44.49 | Dovid | 120* |
| 13:45.32 | leifmadsen | could it be the session-minse setting? |
| 13:45.53 | Dovid | session-expires=91 |
| 13:45.53 | Dovid | session-minse=90 |
| 13:46.04 | Dovid | is it because they r to close together ? |
| 13:46.21 | leifmadsen | I've already exhausted my debugging ability because I just turn them off |
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| 13:46.56 | Dovid | i am lost |
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| 13:52.50 | Dovid | leifmadsen: All I see half way in is: [May 18 09:48:23] DEBUG[16196] chan_sip.c: Session timer expired: 118324 - 24bf41c550c852f55fa650ac35aa6b67@xx.xx.xx.xx |
| 13:55.53 | Dovid | anyone else |
| 13:57.01 | Dovid | ? |
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| 14:08.19 | Khratos | Here goes wwgd again |
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| 14:23.26 | Dovid | \q |
| 14:23.29 | Dovid | quit |
| 14:23.32 | Dovid | quit |
| 14:30.25 | engrxyz | is there any technique for an asterisk clustered asterisk box to be both active-active |
| 14:31.32 | leifmadsen | engrxyz: if you mean active failover and no call loss -- no |
| 14:31.42 | leifmadsen | that's what Asterisk SCF does |
| 14:32.11 | engrxyz | SCF? |
| 14:33.06 | leifmadsen | Scalable Communications Framework |
| 14:33.09 | leifmadsen | google |
| 14:33.19 | leifmadsen | new project started by Digium |
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| 14:39.57 | ruben23 | hi guys how do i cehck if dahdi is working and loaded any idea..? |
| 14:40.53 | Chainsaw | ruben23: dahdi show status |
| 14:41.02 | Chainsaw | ruben23: dahdi show channels |
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| 14:43.46 | ruben23 | Chainsaw: ---> http://pastebin.com/et6Pk9Nk |
| 14:44.40 | WIMPy | Looks like you haven't configured anything. |
| 14:50.27 | Chainsaw | I concur. |
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| 14:56.02 | spetz_ | Any one using Nortel i2002 with chan_unistim |
| 14:58.48 | leifmadsen | nope |
| 14:58.51 | leifmadsen | ~ask |
| 14:58.51 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
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| 15:06.15 | Chainsaw | facepalms |
| 15:06.16 | Chainsaw | 0518160430|so |4|00|[SoDigitMapC]:initDialPlan Maximum of 30 segments reached. Digitmaps ignored: 9,0030xxxxxxxxx|9,0031xxxxxxxxx|9,0033xxxxxxxxx|9,0034xxxxxxxx|9,0045xxxxxxxx|9,0047xxxxxxxx|9,0048xxxxxxxxx|9,00350xxxxxxxx|9,00351xxxxxxxxx|9,00354xxxxxxx|9,00357xxxxxxxx|9,00370xxxxxxxx|9,00371xxxxxxx |
| 15:06.18 | Chainsaw | Le sigh. |
| 15:06.25 | *** join/#asterisk dorphalsig (be939a80@gateway/web/freenode/ip.190.147.154.128) |
| 15:06.31 | dorphalsig | Hello? |
| 15:06.35 | Chainsaw | Yes hi. |
| 15:07.37 | leifmadsen | ohai! |
| 15:07.50 | dorphalsig | I need to upgrade a previous asterisk installation (1.2). |
| 15:08.31 | dorphalsig | and I was wondering if anybody here would have some time to do so |
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| 15:08.33 | dorphalsig | of coufrse |
| 15:08.52 | dorphalsig | it is expected that you charge a fee |
| 15:08.53 | dorphalsig | :) |
| 15:09.53 | spetz_ | I have succesfully got the phone to make inbound and out bound calls but the display does not show the numbers I am dialing |
| 15:10.54 | p3nguin | dorphalsig: Which version do you want to upgrade to? |
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| 15:21.17 | Khratos | About the documentation, where could I find more information about the terms "Family Key, Family, Keytree", etc. |
| 15:21.20 | Khratos | ? |
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| 15:29.01 | leifmadsen | Khratos: perhaps here? http://ofps.oreilly.com/titles/9780596517342/asterisk-DP-Deeper.html#asterisk-CHP-6-SECT-6 |
| 15:29.47 | jaytee | hey! I've got that book! |
| 15:29.56 | leifmadsen | I heard it's ok |
| 15:30.01 | jaytee | chock full o' goodies! |
| 15:30.02 | leifmadsen | authors are dicks though |
| 15:30.05 | jaytee | lol |
| 15:30.09 | leifmadsen | but damn fine looking gentlemen |
| 15:30.29 | mickecarlsson | leifmadsen Chainsaw issue updated https://issues.asterisk.org/view.php?id=18681 |
| 15:30.44 | jaytee | dunno about that, that russell guy looks like he doesn't get enough sleep |
| 15:30.57 | schmidts | leifmadsen i heard one of them is from canada but dont tell ;) |
| 15:31.45 | leifmadsen | mickecarlsson: ok you're closer -- up to 1.6.0.28, but at what point after that is it broken? |
| 15:31.46 | jaytee | I like Canadians... it's their damn geese that emigrated here and now they don't even migrate anymore. they just hang here all the time shitting all over the place. |
| 15:31.55 | leifmadsen | you'll probably have to check 1.6.1.0 to start |
| 15:32.04 | Khratos | Ohhhh.. , thanks leifmadsen |
| 15:32.12 | leifmadsen | jaytee: amen -- some don't fly south anymore and stay here all winter too |
| 15:32.31 | leifmadsen | jaytee: they get lazy, and if they don't fly back for ONE season, they are stuck and can't make the trip for the rest of their lives |
| 15:32.36 | mickecarlsson | leifmadsen: 1.6.1.0 just exits the dialplan and got to 't' exten |
| 15:32.41 | leifmadsen | they lose the muscle mass |
| 15:32.48 | schmidts | :D |
| 15:32.51 | leifmadsen | mickecarlsson: which is correct or incorrect behaviour? |
| 15:32.51 | mickecarlsson | leifmadsen: And that is one bug |
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| 15:33.27 | mickecarlsson | leifmadsen: correct is to go to next exten with ptoper HANGUPCAUSE |
| 15:33.35 | mickecarlsson | proper that is |
| 15:33.36 | leifmadsen | I'm not sure it is.... |
| 15:33.45 | leifmadsen | that sounds like a functionality change that was intentional |
| 15:33.57 | leifmadsen | being able to handle that functionality via the 't' extension is more "correct" |
| 15:34.10 | mickecarlsson | It should NOT exit the dialplan. |
| 15:34.12 | Khratos | leifmadsen: If my memory doesn't fails me (it probably does), I remember you once said something about "personal pages" on the wiki, that might be reviewed by the people in charge, then consider if making them part of the wiki. Did you mentioned something like that? |
| 15:34.33 | leifmadsen | Khratos: I did |
| 15:34.49 | leifmadsen | mickecarlsson: the 't' extension is not exiting the dialplan -- it is still executing in the dialplan |
| 15:35.08 | leifmadsen | it is changing extensions, sure |
| 15:35.18 | leifmadsen | I'm not convinced that isn't the intended behaviour though |
| 15:35.19 | mickecarlsson | Well, IMHO it should go to next exten, and be delt with there |
| 15:35.34 | leifmadsen | so we're of differing opinions |
| 15:35.38 | mickecarlsson | :-) |
| 15:35.47 | leifmadsen | again, you're going to have to track down the commit that changed the behaviour |
| 15:35.49 | mickecarlsson | But why is it 'fixed' in later versions |
| 15:35.56 | mickecarlsson | but with wrong hangupcause |
| 15:35.57 | leifmadsen | that may give some indication as to whether it was intentional or not |
| 15:36.14 | leifmadsen | mickecarlsson: we can debate this all day, but you're going to have to determine which revision specifically changed the functionality |
| 15:36.27 | leifmadsen | only then can we determine whether the change was intentional or not |
| 15:36.32 | mickecarlsson | Ouch, there where over 1000 commits to chan_sip between those verseion |
| 15:36.39 | Khratos | leifmadsen: Is the option "Add Label" in the wiki (above the "Add Comment" option) the one that allows personal pages? |
| 15:36.50 | leifmadsen | Khratos: no, it's called "Personal Pages" :) |
| 15:36.57 | leifmadsen | it's under your name |
| 15:37.08 | leifmadsen | highlight your name in the upper right corner |
| 15:37.56 | mickecarlsson | leifmadsen: then I will have to file two tickets then. One for going to t exten (undocumented) and the present one for giving the wrong hangupcause. sigh. |
| 15:38.05 | leifmadsen | no you don't |
| 15:38.09 | leifmadsen | they are the same issue |
| 15:38.21 | leifmadsen | it sounds like the same commit may have changed that functionality |
| 15:38.35 | leifmadsen | until you find it though, you can't know whether they should be separate issues or not |
| 15:39.44 | mickecarlsson | That is alomst impossible. This is from summary file: channels/chan_sip.c | 3044 ++++++++++++++++++++++------------- |
| 15:39.48 | mickecarlsson | 3044 commits |
| 15:40.17 | Khratos | uuuum... I'm blinder that I thought |
| 15:40.18 | mickecarlsson | For 1.6.1.0 |
| 15:40.22 | leifmadsen | mickecarlsson: it's not impossible if you start jumping large amounts of revisions |
| 15:40.28 | WIMPy | Max 11 tries |
| 15:40.42 | leifmadsen | start at 100, then try 1200, then try 2400, too far? try 1500... still too far? try 1300 |
| 15:40.43 | mickecarlsson | has not touched C code in 20 years |
| 15:41.03 | leifmadsen | many of those commits are going to be merges and other things and not code changes |
| 15:41.04 | WIMPy | Binary tree |
| 15:41.11 | leifmadsen | WIMPy: yay binary tree! :) |
| 15:41.22 | leifmadsen | each actual code change I bet happens only ever 5-8 commits |
| 15:41.34 | leifmadsen | errr... revisions |
| 15:41.43 | mickecarlsson | leifmadsen: where to look? svnview? |
| 15:41.56 | leifmadsen | whatever tool you wish |
| 15:41.58 | leifmadsen | i use svn log |
| 15:42.04 | mickecarlsson | I am only familiar with trac |
| 15:42.17 | mickecarlsson | ok, time to read up |
| 15:42.47 | mickecarlsson | But before I do that, I will test all hangupcauses to se if there was more hickups |
| 15:42.54 | leifmadsen | mickecarlsson: basically, unless you can either resolve the issue yourself, or at least point to the exact revision, your issue is going to get a very low priority |
| 15:43.05 | leifmadsen | which is why I'm trying to help you track down the problem yourself |
| 15:44.13 | Khratos | Could it be that the wiki has user levels, and some of them (in which my user is) does not have the option "Personal Page"? |
| 15:44.21 | WIMPy | I think there are several unfortunate translations for cause codes. |
| 15:45.10 | mickecarlsson | Well, if I look at the code it is documented there, with proper hangupcauses |
| 15:45.27 | mickecarlsson | ok, back to (boring) testing |
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| 16:01.55 | aliverius | is chan_misdn built by default or do i need misdn_utils and such first? also does it make sense to use chan_lcr and lcr itself? |
| 16:02.40 | WIMPy | Yu need misdn_utils. |
| 16:02.50 | WIMPy | And it depends on what you want. |
| 16:03.02 | WIMPy | See http://voice.yeti.dk/Asterisk_vs_ISDN |
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| 16:03.57 | WIMPy | dahdi has ECT as the killer feature. But I personally still prefer LCR. |
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| 16:04.54 | aliverius | so WIMPy is it better to use the old forked misdn or lcr? what i want to do is to access my isdn phone number through the internet |
| 16:05.56 | Qwell | Just use chan_misdn. It's in the Asterisk tree for a reason. |
| 16:06.42 | WIMPy | That would be my last choice. |
| 16:08.03 | *** part/#asterisk freckle (~jon@95.172.10.10) |
| 16:10.04 | aliverius | WIMPy: why? |
| 16:10.48 | WIMPy | Because it only works with quite old Linux versions and I don't see any advantage over dahdi or LCR. |
| 16:12.36 | Qwell | First off, no. |
| 16:12.38 | Qwell | Second, no. |
| 16:12.43 | Qwell | It doesn't use dahdi |
| 16:12.57 | WIMPy | Who said so? |
| 16:13.01 | Qwell | I said so? |
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| 16:13.21 | WIMPy | What does(n't) use dahdi? |
| 16:13.29 | Qwell | misdn, as you're implying |
| 16:13.47 | WIMPy | Where did I do that? |
| 16:14.06 | Qwell | The comment you just made. |
| 16:14.14 | WIMPy | ... and I don't see any advantage over dahdi _or_ LCR. |
| 16:14.25 | Chainsaw | So, country code 1, NANP. |
| 16:14.30 | Qwell | The advantage is that it's in tree. |
| 16:14.30 | Chainsaw | Can I assume fixed-length numbers? |
| 16:14.33 | Qwell | Chainsaw: yes |
| 16:14.46 | Chainsaw | Qwell: Excellent, thanks. |
| 16:14.57 | WIMPy | But both dahdi and LCR offer functional advantages over misdn1. |
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| 16:15.38 | Qwell | If his hardware isn't supported by dahdi, that's kinda moot, don't you think? |
| 16:16.00 | Qwell | as for LCR, yes, the advantage is that chan_misdn is actually supported. |
| 16:16.11 | WIMPy | Yes, but then there's LCR. |
| 16:16.38 | WIMPy | And most hardware can be used with dahdi. |
| 16:16.51 | Qwell | aliverius: What hardware do you have? |
| 16:17.16 | aliverius | Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) |
| 16:17.35 | aliverius | i think this is classic nowadays |
| 16:17.52 | WIMPy | It's bog standard. |
| 16:18.19 | WIMPy | But does indeed need an external module to be used with dahdi. |
| 16:18.31 | WIMPy | external as in 3rd party. |
| 16:18.33 | Qwell | Well, when you convince him to use chan_lcr, he can feel free to go to you for support, I guess. |
| 16:18.46 | aliverius | =/ |
| 16:18.54 | WIMPy | He can :-) |
| 16:19.04 | aliverius | thanks! |
| 16:19.08 | Qwell | Or he can use chan_misdn and report bugs against it, to the Asterisk developers. |
| 16:19.14 | Qwell | just sayin. It's in tree for a reason. |
| 16:19.30 | aliverius | i will try both |
| 16:19.37 | aliverius | so i have questions for both |
| 16:20.09 | aliverius | for chan_misdn i need misdnuser fork? |
| 16:20.25 | aliverius | a kernel patch? |
| 16:20.47 | aliverius | what? |
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| 16:21.15 | WIMPy | A standard kernel up to 2.6.25.4 will do. |
| 16:21.40 | Qwell | I have never heard of such limitation. |
| 16:21.53 | aliverius | my kernel is 2.6.38... |
| 16:22.01 | Qwell | aliverius: https://wiki.asterisk.org/wiki/display/AST/mISDN |
| 16:22.03 | WIMPy | After that version misdn1 was replaced by misdn2. |
| 16:22.04 | aliverius | and cant change for the moment |
| 16:22.45 | aliverius | i have this feeling i am going to use lcr in the end but lets start with chan_misdn |
| 16:23.07 | aliverius | i have installed lcr in the past |
| 16:23.12 | aliverius | didnt know what to do with it lol |
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| 16:26.21 | Khratos | leifmadsen: is here where the "Add Page" should appear? http://slackware-es.com/screenshots/ |
| 16:26.53 | Qwell | Khratos: directly underneath that |
| 16:26.59 | leifmadsen | Khratos: I didn't say anything about Add Page |
| 16:27.03 | Qwell | I do believe pages require edit permission |
| 16:27.10 | Khratos | oohh |
| 16:27.16 | *** join/#asterisk ketema (~ketema@kjhmacpro.ketema.net) |
| 16:27.28 | leifmadsen | Khratos: funny enough I was just asking about the personal spaces, and it seems you need permissions to create one |
| 16:27.43 | leifmadsen | Khratos: first... you ARE logged in right? |
| 16:27.49 | leifmadsen | i.e. you have a user account? |
| 16:27.53 | Qwell | leifmadsen: the name implies so. :) |
| 16:27.53 | Khratos | of course |
| 16:28.39 | leifmadsen | ok I like to check the obvious :) |
| 16:28.43 | Khratos | Given the idiosyncrasy of some users, Madsen question is ok :P |
| 16:28.48 | aliverius | WIMPy: we will stay in touch about lcr, Qwell and #asterisk for misdn |
| 16:28.58 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.lan.noare-1.holmedal.net) |
| 16:29.28 | leifmadsen | Khratos: when I highlight my "Leif Madsen" in the wiki, I get a drop down, and the first thing is "Personal Spaces" |
| 16:29.42 | leifmadsen | I don't think I did anything beyond that but I don't quite remember |
| 16:30.03 | leifmadsen | Khratos: it appears to be a persmissions thing |
| 16:31.48 | Khratos | I thought so, the strange thing is that there's a post suggesting that this might be something available to public ( https://wiki.asterisk.org/wiki/display/TOP/2010/04/15/Create+Personal+Spaces ) |
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| 16:34.34 | *** mode/#asterisk [+o mnicholson] by ChanServ |
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| 16:37.30 | malcolmd | Khratos: log out, log in of confluence; let's see if the change affected on the backend was the correct one :D |
| 16:38.12 | malcolmd | bbiab; lunch |
| 16:39.24 | leifmadsen | me too |
| 16:39.56 | Khratos | malcolmd: still the "Personal Space" does not shows up (logged out/in twice, Firefox cache/cookies deleted) |
| 16:40.12 | leifmadsen | Khratos: I've been told it may take some time for things to sync |
| 16:40.22 | leifmadsen | try again in 10-15 mins or something like that |
| 16:40.36 | Khratos | Ok |
| 16:41.08 | aliverius | Qwell: it seems since i have a new kernel i need not compile mISDN. mISDNuser compiles straightforward. does this mean i am working with v2 which is not compatible with asterisk? what do i do if i have a recent kernel? |
| 16:45.26 | WIMPy | It does. I have never tried to install misdn1 on an misdn2 enabled kernel, but I guess Qwell will tell you more. |
| 16:45.31 | WIMPy | :-) |
| 16:45.38 | *** part/#asterisk gavimobile (~user@bzq-84-108-104-165.cablep.bezeqint.net) |
| 16:45.56 | aliverius | that is what i am waiting for! |
| 16:46.30 | aliverius | (in the meantime lets compile lcr... and then recompile asterisk with for chan_lcr right?) |
| 16:47.23 | WIMPy | You don't recompile Asterisk. chan_lcr is built with LCR. |
| 16:47.28 | WIMPy | But you need the sources. |
| 16:47.42 | WIMPy | Or at least the headers. |
| 16:48.00 | WIMPy | has never installed Asterisk from a package. |
| 16:49.21 | aliverius | with archlinux the asterisk headers are probably there |
| 16:49.52 | aliverius | yes they are |
| 16:50.00 | WIMPy | What Asterisk version? |
| 16:50.05 | aliverius | 1.8.4 |
| 16:50.16 | aliverius | bad choice? |
| 16:50.23 | WIMPy | Ok, you need the asterisk_1_8 branch from git then. |
| 16:50.42 | WIMPy | Or development if you like that. |
| 16:50.45 | aliverius | why? the headers are there |
| 16:50.55 | WIMPy | Of LCR. |
| 16:50.59 | aliverius | ahh |
| 16:51.07 | aliverius | ok |
| 16:51.59 | WIMPy | Seems like noone does tarballs :-(, but the git version works. |
| 17:08.59 | aliverius | is creating a package for misdnuser |
| 17:09.35 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
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| 17:10.59 | *** join/#asterisk wwgd (~wwgd@208.79.14.130) |
| 17:14.03 | wwgd | i recently upgraded to asterisk 1.8.4 on a centos 5.6 system an am trying to upgrade dahdi to 2.1.4.2 but keep getting the following error when i try to compile "You do not appear to have the sources for the 2.6.18-238.9.1.el5PAE kernel installed", anybody know what it means and how I get around it? |
| 17:14.23 | leifmadsen | mean you don't have the sources installed for your running kernal |
| 17:14.41 | leifmadsen | wwgd: sounds like the kernel has been upgraded (packages) but no reboot to make it active |
| 17:14.47 | leifmadsen | so the sources are out of sync |
| 17:15.25 | leifmadsen | try: yum install kernel-devel-`uname -r` perhaps |
| 17:15.28 | wwgd | hmmm, ok, i did do a yum upgrade recently, maybe that's it, thanks |
| 17:15.28 | *** join/#asterisk Denial (~Denial@drgi.co.uk) |
| 17:15.33 | leifmadsen | that'd be the problem then |
| 17:15.40 | leifmadsen | the sources don't match the running kernel |
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| 17:33.40 | leifmadsen | wwgd: is it your connection or your IRC software that causes you to drop constantly? |
| 17:34.39 | wwgd | my connection, i'm out at the end of a satellite shot and there is a sandstorm, i'm in iraq |
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| 17:35.21 | leifmadsen | gotcha |
| 17:35.32 | wwgd | i'm trying the install now for the kernel that you suggested, we'll see if it works |
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| 17:36.04 | wwgd | appreciate the advice, you really seem to know your way around this system |
| 17:36.51 | leifmadsen | I know only 5 things, and you picked one of them |
| 17:36.54 | leifmadsen | burn |
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| 17:47.53 | quid246 | Is there a way to write text to a file from 1.6.2 other than the System() command? |
| 17:49.50 | leifmadsen | quid246: use SHELL() ? |
| 17:51.10 | quid246 | Hmm... not familiar with it, let me check up on it. |
| 17:51.16 | leifmadsen | core show function SHELL |
| 17:52.21 | quid246 | I suppose it would work... I guess I was hoping for something more specific rather than a swiss army knife. Thought I saw a patch someone submitted that would allow writes in 1.6 but I guess it never made it into the branch. |
| 17:52.25 | *** join/#asterisk twouters (~twouters@unaffiliated/twouters) |
| 17:52.36 | leifmadsen | quid246: sounds like you want an AGI() |
| 17:52.42 | *** join/#asterisk axilla (~axilla@67.238.59.34) |
| 17:52.54 | quid246 | I dislike SYSTEM for the fact it can be a security hole... as opposed to a more specific WRITE function. |
| 17:53.15 | axilla | anyone know of a good way to bill inbound customers.. say for a law firm that has customer calling in and are on billable time? |
| 17:53.16 | leifmadsen | doesn't sound like something that should be handled by asterisk directly |
| 17:53.33 | leifmadsen | axilla: write information to a database? use CDRs? |
| 17:53.38 | axilla | yea |
| 17:53.43 | leifmadsen | have at it |
| 17:53.48 | axilla | i'm aware of that, but just checking to see if i need to reinvent the weheel. |
| 17:53.57 | leifmadsen | I don't think your question is specific enough |
| 17:53.59 | axilla | or if there is something opensource out there. |
| 17:54.10 | axilla | i need to be able to do inbound call billing.. thats pretty much it. |
| 17:54.10 | *** part/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:54.46 | axilla | some type of billing integration for inbound calls when a client calls a lawyer and a lawyer wants to charge them for the time on the call. |
| 17:54.54 | quid246 | I was thinking of that. Basically if * can't post a CDR/account update via ODBC (connection down or what not), want it to write the information to a unique file per call. Then just have a cron job that runs every 15 mintues or so, retries the DB and deletes each file upon success. |
| 17:56.33 | *** join/#asterisk wwgd (~wwgd@208.79.14.130) |
| 17:58.57 | wdoekes2 | quid246: wouldn't it make more sense to write to a textual cdr.. and delegate the cdr.csv parsing, rotating and writing to db to a different process? |
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| 18:00.52 | wdoekes2 | if my db fails me then everything stops working. so having only cdr writes offloaded to text makes no sense to me at all |
| 18:01.03 | quid246 | wdoekes2: Probably. Perhaps I could write to a text file via ODBC... I haven't used CSV based CDRs in years. |
| 18:03.12 | mickecarlsson | leifmadsen ping |
| 18:05.00 | zkn | has anyone installed asterisk on amazon cloud lately? |
| 18:08.04 | zkn | all looks genreally good but whatever I try i cannot get sound working for calls...every call ends with Retransmission timeout reached on transmission |
| 18:08.33 | zkn | nat=yes or no, no difference |
| 18:09.53 | quid246 | Security groups setup properly... externip/externhost and localnet in sip.conf set properly? |
| 18:10.06 | zkn | that was it |
| 18:10.15 | zkn | just noticed a type in externip |
| 18:10.19 | zkn | jeesh |
| 18:10.20 | quid246 | cool |
| 18:10.25 | zkn | typo* |
| 18:12.28 | *** join/#asterisk Dibbler (~Dibbler@87-194-103-72.bethere.co.uk) |
| 18:13.29 | *** join/#asterisk Dibbler (~Dibbler@87-194-103-72.bethere.co.uk) |
| 18:13.50 | zkn | ok, thanks for all the help, signing out |
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| 18:21.03 | *** join/#asterisk nny (~Scott_2@174.107.201.103) |
| 18:21.07 | nny | ahh.. provider issues |
| 18:21.10 | nny | how i love thee |
| 18:35.35 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 18:36.44 | *** join/#asterisk Syre_x (~syrex@dsl-146-17-172.telkomadsl.co.za) |
| 18:38.04 | nny | anyone see a problem here? |
| 18:38.05 | nny | http://min.us/mbjBvWPAvUpsof |
| 18:38.13 | nny | i see only one rtp stream |
| 18:38.31 | nny | 10.10 is provider, 64 is local fyi |
| 18:40.00 | ferdna | guys when ever people call me they tell me they don't hear a ringtone... |
| 18:40.07 | ferdna | but i can see the call come in |
| 18:40.57 | *** join/#asterisk Failrar (~Failrar@5ED66E6D.cm-7-7b.dynamic.ziggo.nl) |
| 18:41.15 | ferdna | and i can answer the call just fine |
| 18:41.52 | nny | try adding ring to your dial statement |
| 18:42.01 | ferdna | nny, what command is that? |
| 18:42.05 | nny | dial |
| 18:42.12 | ferdna | i have it in there |
| 18:42.22 | WIMPy | Ringing() |
| 18:42.28 | Qwell | Are you calling Answer() before Dial()? |
| 18:42.43 | WIMPy | Or what was there? |
| 18:44.40 | ferdna | nny, WIMPy, Qwell: http://pastebin.com/GPydkxid |
| 18:44.48 | *** part/#asterisk quid246 (~quid249@CPE00131078b0b5-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 18:44.58 | Qwell | ferdna: Get rid of the answer |
| 18:45.42 | ferdna | awesome thanks =) |
| 18:53.51 | pushpop | I havent checked in a while, does asterisk have the ability to have the one extension that is registered on multiple phones? |
| 18:54.58 | WIMPy | no |
| 18:55.25 | pushpop | rgr |
| 18:56.56 | Qwell | why not? |
| 18:57.30 | Qwell | An extension does whatever you tell it to do. If you want it to dial multiple phones, then do so. |
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| 19:03.03 | nny | help. I have a trunk that uses it's own interface |
| 19:03.14 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:03.24 | nny | how do I tell asterisk that this trunk should only send public ip dialog on it's interface |
| 19:03.30 | nny | nat=no should set it right? |
| 19:04.19 | nny | Contact: <sip:8005421048@70.167.35.228:5060> |
| 19:04.21 | nny | is the issue |
| 19:04.24 | nny | it should be different |
| 19:13.15 | nny | http://pastebin.com/Q1L456FJ |
| 19:13.19 | nny | is what I am seeing |
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| 19:29.31 | *** join/#asterisk fullstop (~fullstop@static-173-210-91-4.saucontech.com) |
| 19:29.56 | fullstop | Howdy. I'm looking for some advice... and I hope that it's not "DON'T DO IT!!" |
| 19:30.35 | fullstop | I need to get fax working over SIP. I'll be talking to our ITSP shortly and hopefully we're going to try T.38. |
| 19:31.42 | fullstop | In the mean time, I'd like to have g711 as a backup. Their codec selection is g729/ulaw in that order. Is it possible for me to accept the calls in ulaw and fax detect and then switch to g729 or am I taking the wrong approach? |
| 19:32.08 | fullstop | Obviously, I mean switch to g729 if it is not a fax. |
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| 19:35.11 | *** join/#asterisk Defraz (~Defraz@63.226.95.152) |
| 19:35.49 | nny | ok need some help with this issue: Our provider is sending their end of the RTP stream to the wrong interface on our network. This is obviously bad and breaking things. I see contact:inboundnumber@wrongip (this only breaks on inbound, outbound works fine). Need some help diagnosing further |
| 19:37.02 | fullstop | Where are they getting that ip address from? |
| 19:37.33 | nny | fullstop: great question |
| 19:37.38 | nny | http://pastebin.com/Q1L456FJ |
| 19:37.46 | fullstop | are you setting udpbindaddr in sip.conf? |
| 19:38.11 | nny | fullstop this is specific to one trunk |
| 19:38.20 | nny | fullstop: can it be set under the per definition? |
| 19:38.22 | nny | peer* |
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| 19:40.01 | fullstop | I am pretty sure that it is [general] only. |
| 19:40.28 | fullstop | nny: This would be a bandaid, but can you redirect it w/iptables? |
| 19:41.50 | nny | fullstop: no |
| 19:41.58 | nny | fullstop: the provider is sending it to the wrong box/ip |
| 19:42.19 | fullstop | For some reason I thought that both interfaces were on the same server. |
| 19:42.26 | nny | fullstop: er they are sorry nm |
| 19:42.48 | nny | fullstop: but yeah maybe. I am more concerned why the dialog shows Contact: <sip:NUMBER@WRONGIP:5060> |
| 19:43.13 | nny | nat=no is set, the box is sending it's dialog to the provider on the proper ip |
| 19:43.25 | nny | trying to figure out why that contact field is flat out wrong |
| 19:43.45 | fullstop | I'm not terribly familiar with how the various nat settings in sip.conf change things.. |
| 19:43.52 | fullstop | but is externip set? |
| 19:44.09 | fullstop | or is 70.167.35.228 the first interface on the server? |
| 19:44.44 | fullstop | I'm just trying to understand why it picks that address / where it gets it from. |
| 19:45.47 | nny | fullstop: yes, but nat=no in peer definition |
| 19:45.50 | nny | fullstop: yeah me too |
| 19:46.16 | nny | fullstop: nat=no, it shouldn't change it's dialog to match externip |
| 20:03.35 | jaytee | I have an interesting problem. I have set my iptables firewall to accept 5060 udp, 10000-20000 rtp and 4569 for IAX2. When I have the iptables service started and make a test call outbound to my cell I get no audio in both directions and when I stop the service and make a test call I get audio bidirectional. I have udp for RTP 10000:20000 set to accept in my input chain. I'm running CentOS. |
| 20:06.10 | jaytee | and my * box is behind a nat'd firewall/router and I have nat=yes for the peer definition of my sip provider. |
| 20:06.36 | fullstop | a firewall and iptables. That's like being behind 7 proxies. |
| 20:07.08 | jaytee | yeah, I should just disable iptables and leave it all up to the firewall |
| 20:07.44 | fullstop | If you want to continue to use iptables, make a call and do iptables -vnL and see if your byte counters are increasing. |
| 20:07.58 | fullstop | If not, your iptables rules are probably incorrect. |
| 20:12.44 | fullstop | sip show peer PEERNAME -- does "T.38 Support : Yes" indicate what is in sip.conf or what the peer actually supports? |
| 20:13.10 | mickecarlsson | leifmadsen I have found the two bugs for the 484 |
| 20:14.16 | *** join/#asterisk hugogee (~hugogee@cpe-76-175-209-64.socal.res.rr.com) |
| 20:14.21 | hugogee | greets all :D |
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| 20:25.40 | mickecarlsson | leifmadsen Chainsaw issue updated, now it is up to you (with a little help from me) https://issues.asterisk.org/view.php?id=18681 |
| 20:26.30 | mickecarlsson | I have tested it on Asterisk 1.8.4 with the removal of the code, and now 1.8.4 return the correct hangupcasew |
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| 20:27.52 | mickecarlsson | gtg, bedtime in Sweden |
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| 20:31.21 | fullstop | It's only 10:30 there. That's far too early to go to bed. |
| 20:33.52 | mickecarlsson | LOL |
| 20:34.11 | mickecarlsson | In Malmo it is way past bedtime |
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| 20:37.54 | _pll | Hi, does anyone know of a command line tool to compare two audio files? |
| 20:38.52 | Aut0ExeC | like compare and show you whats diff? |
| 20:39.09 | Aut0ExeC | wonder how that would be possible |
| 20:39.28 | _pll | Not really, the percentage of difference is enough. |
| 20:39.34 | Aut0ExeC | oh |
| 20:39.47 | _pll | I am trying to find certain tones at the beginning of a file. |
| 20:39.53 | Aut0ExeC | i dunno bro but... audacity came to mind for gui |
| 20:39.57 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 20:40.05 | _pll | They aren't dtmf tones. |
| 20:40.13 | Aut0ExeC | oh ok |
| 20:40.22 | _pll | I need to automatice this, preferable without gui. |
| 20:40.26 | Aut0ExeC | k |
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| 20:43.35 | _pll | I was thinking in using the fast fourier transform of the waves and compare them but I think this is common enough to have a tool somewhere in the net. |
| 20:44.07 | Aut0ExeC | yah oh man.... its hard enough to find a tool... much less a cli one |
| 20:44.46 | _pll | Yeah, google has failed me. That's why I am here. |
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| 21:14.45 | ferdna | guys what is wrong with that warning? |
| 21:14.46 | ferdna | http://pastebin.com/G1G1yyUW |
| 21:14.48 | *** part/#asterisk Aut0ExeC (~Jack@24.244.156.75) |
| 21:17.28 | fullstop | ferdna: your dialplan is messed up. |
| 21:17.54 | fullstop | each extension has to start with 1 and the following lines can have "n" for the number. |
| 21:19.59 | ferdna | fullstop, http://pastebin.com/iX3wwFQJ |
| 21:21.25 | ferdna | fullstop, oh yeah... i changed to one and now i dont get that message |
| 21:23.36 | ferdna | thanks |
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| 22:05.21 | nny | what's the preferred method for polycom mwi? contact? |
| 22:06.28 | ArtemMakhutov | Is there any chance to edit the description and title of a bug after opening it on issues.asterisk.org ? |
| 22:09.02 | bbryant | ArtemMakhutov: I think you have to be an admin or moderator for that. Leave a comment detailing what you would like to change, and send me a PM with the bug number. |
| 22:09.20 | bbryant | that way there's a record indicating why I changed something on someone else's bug report |
| 22:10.16 | ArtemMakhutov | ok, thx |
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| 22:23.06 | *** join/#asterisk bmoraca (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 22:23.33 | bmoraca | i'm looking for an E911 provider...anyone have any recommendations? |
| 22:35.31 | carrar | bmoraca, www.911enable.com |
| 22:36.03 | Freeaq | < european guy here. What's e911? |
| 22:36.15 | Freeaq | I thought it was just a directory of number vs location? |
| 22:38.27 | *** join/#asterisk WiretapWork_ (~Wiretap@unaffiliated/wiretap) |
| 22:40.48 | *** join/#asterisk aliverius (~quassel@athedsl-376932.home.otenet.gr) |
| 22:41.12 | bmoraca | carrar: are they competitive for a smaller provider? |
| 22:42.05 | bmoraca | dash (bandwidth.com) has a $1000 monthly minimum |
| 22:42.19 | bmoraca | that's pretty ridiculous, as we don't have 1000 numbers that need 911 |
| 22:55.53 | WiretapWork_ | bmoraca, seriously, its not the law for a provider to provide 911 connection over there? |
| 22:56.09 | bmoraca | WiretapWork_: it is |
| 22:56.26 | WiretapWork_ | then why do you have to pay to have 911? |
| 22:56.36 | bmoraca | WiretapWork_: my existing wholesale provider does 911...the one we're looking at moving to does not provide 911 services for wholesale accounts |
| 22:56.44 | WiretapWork_ | what the shit |
| 22:56.50 | bmoraca | as such, I need a 3rd party 911 provider |
| 22:56.51 | WiretapWork_ | see, that would be illegal here |
| 22:57.13 | bmoraca | why? the company doesn't provide end-user phone service...thus no need for 911. |
| 22:57.34 | WiretapWork_ | any company that provides telecommunications service of any kind here has to provide 911 |
| 22:57.38 | WiretapWork_ | or, rather, 111 |
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| 22:58.05 | bmoraca | even in wholesale? that's kind of silly |
| 22:58.39 | bmoraca | and redundant |
| 22:59.17 | WiretapWork_ | is 911 not centralised to a government run callcentrethere or something? |
| 22:59.34 | WiretapWork_ | here it is just trunked back to a government-owned-and-run callcentre |
| 23:00.41 | bmoraca | there are local PSAPs...the telco is required to route the call to the appropriate PSAP based on the location of the caller and also to provide address and callback number information |
| 23:01.53 | WiretapWork_ | yep, exactly the same as here |
| 23:02.47 | bmoraca | i don't have the capability to do that myself, which is why I need a 3rd party to do it. my wholesaler does it for me now, but the future one does not currently do that |
| 23:05.06 | *** part/#asterisk ArtemMakhutov (~ArtemMakh@ip-95-223-4-6.unitymediagroup.de) |
| 23:07.37 | *** join/#asterisk michael-i (~michael@204.11.230.58.static.etheric.net) |
| 23:08.41 | michael-i | Hi all. I'm trying to get ChanSpy working on a 1.8.3.2 system dealing only with SIP channels. directmedia=no is set and the calls are bridged but when I dial into the ChanSpy extension, the bridged call is destroyed/hungup. |
| 23:08.55 | michael-i | Do I need any additional res_ or bridge_ bits loaded to accomplish this? |
| 23:11.26 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 23:12.48 | michael-i | Looks like I might have found an already reported bug: https://issues.asterisk.org/view.php?id=18647 |
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| 23:34.00 | sawgood | using a command like sip show peer xxx (is there anyway to tell if the SIP end point has its DND service active)? |
| 23:34.12 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 23:34.17 | sawgood | Or, is there another way to learn this if you are remote to the phone? |
| 23:37.37 | aliverius | WIMPy: i need lcr from git for asterisk 1.8 right? |
| 23:37.51 | aliverius | is 1.8 worth it? |
| 23:38.53 | aliverius | and where is that git? or svn? or whatever? |
| 23:40.06 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 23:41.59 | aliverius | git found |
| 23:43.46 | WIMPy | aliverius: yes. te "asterisk_1_8" branch. |
| 23:43.55 | WIMPy | And yes, I'd use 1.8. |
| 23:44.03 | aliverius | git doesnt build |
| 23:44.06 | aliverius | :( |
| 23:44.31 | WIMPy | Did you change the branch? |
| 23:46.09 | WIMPy | There's that script to do so. |
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| 23:53.00 | aliverius | WIMPy: an option in the configure script |
| 23:53.06 | aliverius | cause i cant find anythign else |
| 23:53.51 | WIMPy | sh checkout-branch.sh asterisk_1_8 |
| 23:54.47 | aliverius | WIMPy: did God send you? |
| 23:54.55 | WIMPy | no |
| 23:54.58 | aliverius | i have tried to do stuff with isdn in the past |
| 23:55.05 | aliverius | never found the help i needed |
| 23:55.28 | WIMPy | I know, I should write a howto, but didn't get that far, yet. |
| 23:55.42 | aliverius | are you a ev? |
| 23:55.43 | aliverius | dev? |
| 23:56.12 | WIMPy | What do you have to to to qualify as such? |
| 23:56.21 | WIMPy | I submitted some patches. |
| 23:56.33 | WIMPy | But that's it. |
| 23:57.16 | aliverius | then you are experienced enough :D |
| 23:57.48 | WIMPy | Andreas did LCR more or less on his own. |
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| 23:58.22 | aliverius | andresmujica? |
| 23:58.41 | WIMPy | Andreas Eversberg |
| 23:58.48 | michael-i | Just popping back in to say that the bug I reported a bit ago was solved by upgrading to 1.8.4. No more violent chanspy kicks. That is all... |
| 23:59.41 | aliverius | now i have to package lcr too and start playing :) |