| 00:05.57 | navaismo | and the latest remove some features :'( |
| 00:14.40 | file | navaismo, ummm how so? |
| 00:15.16 | navaismo | no more macro variables using AEL |
| 00:17.54 | *** join/#asterisk bkruse (~Adium@24.42.207.11) |
| 00:19.00 | navaismo | well go to go |
| 00:19.20 | *** join/#asterisk bmg505 (~leon@196-215-37-67.dynamic.isadsl.co.za) |
| 00:20.19 | [TK]D-Fender | file: To elaborate, AEL Macro's now parse to Gosub's instead and you no longer get MACRO_EXTEN because of it. |
| 00:20.57 | [TK]D-Fender | file: which is a misnomer now and a loss of a piece of previous functionality |
| 00:21.39 | file | it's done that for years O.o |
| 00:22.11 | file | but, indeed |
| 00:22.31 | [TK]D-Fender | Was a surprise to him... I can't confirm how long ago it changed, just that what it was doing.... |
| 00:22.49 | [TK]D-Fender | I don't waste my time with middleware :) |
| 00:22.55 | file | let's take a gander |
| 00:30.05 | file | huh |
| 00:31.46 | *** join/#asterisk hfp (~hfp@MTRLPQ0736W-LP130-01-2925269581.dsl.bell.ca) |
| 00:32.17 | file | I get lost in the noise... |
| 00:33.47 | hfp | Hi, I'm using Asterisk 1.8.13 on Debian Wheezy. I moved my Asterisk to a virtual machine. Everything works fine except a website that requests a call via the AMI. It is supposed to be TLS encrypted. When I telnet to the AMI host and port, I can connect without encryption. If I use `openssl s_client` then I am getting an error "SSL23_GET_SERVER_HELLO:unknown protocol". I suspect the problem to be on the Asterisk machine |
| 00:33.48 | hfp | because nothing has changed on the client's side. Where should I look to troubleshoot this? |
| 00:34.54 | file | are you connecting to the right port? the default config has TLS encrypted AMI on 5039 |
| 00:36.03 | hfp | Ah that might be why |
| 00:36.14 | hfp | I was using 5038 for my tests |
| 00:38.11 | *** part/#asterisk mobiusnz (~mobiusnz@116.90.137.122) |
| 00:39.46 | hfp | On port 5039 I am getting a connection refused error. Ports are open and forwarded all the way to the asterisk, how can I check if the AMI is configured right? |
| 00:40.08 | hfp | It's a TCP connection on port 5039, right? |
| 00:40.39 | TazzNZ | hfp: on the asterisk box, run "netstat -nvlp | grep asterisk" |
| 00:41.06 | file | check manager.conf, and make sure that the cert and private key file it points to are also present |
| 00:42.18 | hfp | TazzNZ: It shows ports 5038, 2000, 4569, 5000, 4520 and 5060 |
| 00:42.31 | TazzNZ | then see what file said :) |
| 00:43.06 | hfp | file: Yes the path is right. Which user does the asterisk daemon runs as? Maybe it's a permission issue? |
| 00:43.35 | TazzNZ | hfp: normally asterisk:asterisk |
| 00:43.40 | TazzNZ | under Deb/Ubuntu |
| 00:44.13 | TazzNZ | and even CentOS/Redhat it appears |
| 00:44.23 | TazzNZ | unless that is AsteriskNOW doing something funky |
| 00:45.03 | hfp | After changing manager.conf do I have to restart the daemon? Or can I just reload AMI from the console? |
| 00:45.12 | TazzNZ | but you could run "ps -Af | grep asterisk | grep -c" to find out |
| 00:45.30 | TazzNZ | manager reload should do it from the CLI |
| 00:49.40 | hfp | Ok so it seems to be a permission error on the .key file. Right now it is owned by root:root and set to 400. What do you recommend? |
| 00:49.47 | hfp | The .crt file is 644 root:root |
| 00:50.00 | TazzNZ | asterisk:asterisk |
| 00:50.04 | hfp | should I just chown them to asterisk:asterisk? |
| 00:50.05 | TazzNZ | and 400 for both |
| 00:50.07 | hfp | ok |
| 00:51.30 | *** join/#asterisk pcsphone (~root@c-76-105-57-202.hsd1.ca.comcast.net) |
| 00:52.11 | hfp | It works now, thanks TazzNZ and file |
| 00:52.55 | pcsphone | Q: please msg me if you're a consultant. |
| 00:54.01 | WIMPy | pcsphone: If you're looking for someone to help you, you need to tell what you need help with. |
| 00:54.27 | TazzNZ | np hfp |
| 00:55.22 | pcsphone | i'm looking to replace a propratory pbx system with asterisk. i need a sip/voip phone with at least 12 programable buttons and help setting up those buttongs to page extensions. |
| 00:56.49 | TazzNZ | pcsphone: the general channel will help with Asterisk questions |
| 00:57.25 | hfp | TazzNZ: Another question, should I protect my AMI with fail2ban? |
| 00:57.49 | TazzNZ | hfp - you are running that behind a firewall/NAT ? |
| 00:57.59 | WIMPy | hfp: You most probably shouldn;t have it reachable at all. |
| 00:58.00 | hfp | TazzNZ: Yes |
| 00:58.18 | hfp | Well it's reachable only from the IP the webserver requesting the callback runs on |
| 00:58.23 | TazzNZ | yeah - that is where I was heading :D |
| 00:58.49 | TazzNZ | in which case, only if they hijack the IP can they get in |
| 00:59.00 | TazzNZ | if they crack the web server then you have other problems on your hands |
| 00:59.06 | TazzNZ | as they will have most of the passwords |
| 00:59.09 | WIMPy | It's not on your net? |
| 00:59.33 | hfp | WIMPy: The Asterisk runs one my home server but the webserver requesting the callbacks is hosted |
| 00:59.44 | hfp | on* |
| 01:00.17 | WIMPy | Should be fine as long as it's only opened for that IP. |
| 01:00.34 | pcsphone | -> *nickserv* identify asdfasdf |
| 01:00.34 | pcsphone | -NickServ- You are already logged in as pcsphone. |
| 01:00.34 | pcsphone | -NickServ- Please check your email for instructions to complete your |
| 01:00.34 | pcsphone | +registration. |
| 01:00.34 | pcsphone | -> *NickServ* VERIFY REGISTER pcsphone ncxcyjsivbey |
| 01:00.34 | pcsphone | -NickServ- pcsphone has now been verified. |
| 01:00.35 | pcsphone | -NickServ- Thank you for verifying your e-mail address! You have taken steps |
| 01:00.35 | pcsphone | +in ensuring that your registrations are not exploited. |
| 01:00.36 | pcsphone | -> *nickserv* identify asdfasdf |
| 01:00.36 | pcsphone | -NickServ- You are already logged in as pcsphone. |
| 01:00.37 | pcsphone | *** pcsphone (~root@c-76-105-57-202.hsd1.ca.comcast.net) has joined channel |
| 01:00.37 | pcsphone | +#asterisk |
| 01:00.38 | pcsphone | *** Topic for #asterisk: #asterisk The Open Source PBX and Telephony Platform |
| 01:00.38 | pcsphone | +(asterisk.org) -=- LTS: Asterisk 11.9.0 (2014/04/23), 1.8.27.0 (2014/04/23); |
| 01:00.39 | pcsphone | +Standard: Asterisk 12.2.0 (2014/04/23); DAHDI: DAHDI-linux 2.9.0 |
| 01:00.39 | pcsphone | +(2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.14 (2012/12/20) |
| 01:00.42 | hfp | Yes, I forgot I set that in the fw... TazzNZ's question re nat just reminded me |
| 01:00.52 | WIMPy | How secure your web server is, is beyound our topic :-) |
| 01:00.53 | hfp | pcsphone: wtf |
| 01:01.01 | pcsphone | misclick |
| 01:01.03 | pcsphone | sorry |
| 01:01.12 | hfp | WIMPy: haha yes indeed |
| 01:01.37 | hfp | pcsphone: 'asdfasdf' is a bad password :) |
| 01:01.47 | pcsphone | hfp: hahaha |
| 01:01.54 | TazzNZ | lol |
| 01:02.04 | TazzNZ | and IRC'ing as root..... |
| 01:02.09 | hfp | and running your irc client as root not the best idea either |
| 01:02.35 | pcsphone | hfp: this is just a temp acccount. |
| 01:02.50 | hfp | pcsphone: cool |
| 01:08.38 | *** join/#asterisk JuStIcIa_ (~JuStIcIa_@186.6.243.241) |
| 01:09.19 | *** join/#asterisk bmurt (~brendan@208-58-116-232.c3-0.upd-ubr1.trpr-upd.pa.cable.rcn.com) |
| 01:12.50 | *** join/#asterisk theron (~theron@ip66-43-220-25.static.ishsi.com) |
| 01:13.21 | *** join/#asterisk vinhdizzo (~vinh@cpe-98-154-210-61.socal.res.rr.com) |
| 01:17.42 | *** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au) |
| 01:32.56 | *** join/#asterisk theron (~theron@ip66-43-220-25.static.ishsi.com) |
| 01:42.52 | *** join/#asterisk timahvo1 (~rogue@197.237.131.169) |
| 01:43.20 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 01:56.40 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 02:09.18 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 02:25.36 | *** join/#asterisk bkruse (~Adium@24.42.207.11) |
| 02:34.54 | *** join/#asterisk leifmadsen (~free-mads@asterisk/documenteur-extraordinaire/blitzrage) |
| 02:34.54 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 02:51.13 | *** join/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
| 02:51.56 | *** join/#asterisk LiuYan (~LiuYan@211.162.64.249) |
| 03:10.51 | *** join/#asterisk shidan (~chatzilla@CPEc0c1c0c0a9ae-CM001371871af0.cpe.net.cable.rogers.com) |
| 03:46.08 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-56-12.rn.hr.cox.net) |
| 03:59.38 | *** join/#asterisk raspberrypifan (~textual@71-22-220-224.gar.clearwire-wmx.net) |
| 04:03.56 | *** join/#asterisk wolrah (~wolrah@24.239.210.140) |
| 04:25.13 | *** join/#asterisk vinhdizzo (~vinh@cpe-98-154-210-61.socal.res.rr.com) |
| 04:29.15 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 04:38.39 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
| 05:09.54 | *** join/#asterisk spicyramen_ (~Adium@c-98-210-160-181.hsd1.ca.comcast.net) |
| 05:10.44 | spicyramen_ | ciao anyone has deployed asterisk newer version with HD video phones (720p video quality), if so which phone is recommended? |
| 05:12.34 | snadge | fritzbox 7390.. its not accepting incoming calls and I'm struggling to find out why |
| 05:13.57 | snadge | sending device rings for a few rings.. then hangs up busy.. receiving device doesnt ring at all. .the only clue in the sip debug that I can see |
| 05:14.04 | snadge | SIP/2.0 480 Temporarily not available |
| 05:30.13 | *** join/#asterisk AlHafoudh (~textual@dsl-static-173.212-5-200.telecom.sk) |
| 05:32.03 | *** join/#asterisk CeBe (~CeBe@port-92-206-19-45.dynamic.qsc.de) |
| 05:35.32 | *** join/#asterisk timahvo1 (~rogue@197.237.131.169) |
| 06:06.32 | *** join/#asterisk woopstar (~woopstar@unaffiliated/woopstar) |
| 06:08.40 | *** join/#asterisk bkruse (~Adium@24.42.207.11) |
| 06:22.07 | *** join/#asterisk bulkorok (~Adium@85.183.61.47) |
| 06:30.07 | woopstar | Is there something special to do to get WebRTC to work in Asterisk 11.9? I enabled the https server (and its bound to 8089). Created TLS certificate as written in the files. But SipML5 cannot connect to a WebSocket Secure. |
| 06:36.30 | *** join/#asterisk r00f (~r00f@bba166468.alshamil.net.ae) |
| 06:58.09 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 07:00.31 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:307e:45b9:ac41:5681) |
| 07:13.57 | *** join/#asterisk yago (~kresp0@gateway/tor-sasl/kresp0) |
| 07:26.05 | *** join/#asterisk AlHafoud_ (~textual@85.248.11.120) |
| 07:32.15 | *** join/#asterisk Zogot (~Adium@D4B2620B.static.ziggozakelijk.nl) |
| 07:33.12 | Zogot | Ahoyhoy |
| 07:45.37 | *** join/#asterisk jhlavacek (~jirka@80.82.225.246) |
| 07:46.40 | *** join/#asterisk BakaKuna (~Thunderbi@office.voys.nl) |
| 08:01.13 | TazzNZ | snadge: you need to at least say how you connecting them |
| 08:13.57 | andi | Hi |
| 08:14.07 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 08:14.17 | andi | Can somebody tell me how a OPTIONS packet looks like in the asterisk console? |
| 08:14.32 | snadge | yeah.. im off work now, but I did some googling and apparently there are issues with fritz 7390 and inbound calls, codec related |
| 08:15.12 | *** join/#asterisk DelphiWorld (~TheVoicer@openvpn/user/DelphiWorld) |
| 08:15.15 | DelphiWorld | hi guys |
| 08:15.27 | DelphiWorld | anyone done a call button using webrtc? |
| 08:15.57 | snadge | uniden handset dect.. plugged into the analog port via its base station, apparently theres issues with them not ringing as well |
| 08:18.11 | snadge | apparently uniden handsets are cringe inducing to the voip support techs that i've mentioned it to |
| 08:27.40 | *** join/#asterisk r00f (~r00f@94.200.97.245) |
| 08:42.14 | *** join/#asterisk Gugge (gugge@kriminel.dk) |
| 08:50.55 | *** join/#asterisk _omer (~omer@116.71.190.163) |
| 08:52.06 | _omer | hello, How can I create a custom cdr-csv file? or if I can save "Noop(---whatever I type here---)" in cdr-csv file? I need to Log my "Noop()" somewhere in the file ... any help? |
| 09:03.59 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 09:06.57 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 09:09.03 | *** join/#asterisk woopstar (~woopstar@unaffiliated/woopstar) |
| 09:09.27 | woopstar | Is there something special to do to get WebRTC to work in Asterisk 11.9? I enabled the https server (and its bound to 8089). Created TLS certificate as written in the files. But it seems SipML5 is not able to connect at all. |
| 09:10.01 | *** join/#asterisk jhlavacek (~jirka@80.82.225.246) |
| 09:10.06 | *** part/#asterisk at0mat (~at0mat@d216-232-158-23.bchsia.telus.net) |
| 09:10.12 | *** part/#asterisk StaRetji (~vencuga@178.79.6.17) |
| 09:17.08 | *** join/#asterisk CeBe (~CeBe@port-92-206-19-45.dynamic.qsc.de) |
| 09:24.41 | *** join/#asterisk 77CAALUOX (~Thunderbi@cpc5-nott16-2-0-cust33.12-2.cable.virginm.net) |
| 09:27.41 | *** join/#asterisk af_ (~af@93-43-45-195.ip90.fastwebnet.it) |
| 09:30.55 | snadge | oh what, apparently theres an incoming call type setting.. which can be A, B, C or D.. and a privacy/call blocking setting.. on uniden handsets |
| 09:32.57 | *** join/#asterisk jhlavacek (~jirka@80.82.225.246) |
| 09:35.03 | *** part/#asterisk CeBe (~CeBe@port-92-206-19-45.dynamic.qsc.de) |
| 09:43.30 | *** join/#asterisk DelphiWorld (~TheVoicer@openvpn/user/DelphiWorld) |
| 09:45.57 | DelphiWorld | guys |
| 09:46.04 | DelphiWorld | no one done any click to call button for webrtc? |
| 09:46.10 | DelphiWorld | using jssip or sip.js |
| 09:52.02 | *** join/#asterisk jhlavacek (~jirka@80.82.225.246) |
| 09:55.01 | TazzNZ | DelphiWorld: yes - I have |
| 09:55.14 | DelphiWorld | TazzNZ: what sip stack? |
| 09:55.50 | TazzNZ | uhm |
| 09:55.56 | TazzNZ | let me recall.... |
| 10:00.49 | TazzNZ | I think it was jssip |
| 10:00.56 | DelphiWorld | jssip is cleaner |
| 10:01.58 | TazzNZ | you know that webrtc needs srtp |
| 10:02.33 | DelphiWorld | yep sure |
| 10:02.38 | DelphiWorld | but not on remote side:) |
| 10:02.48 | DelphiWorld | TazzNZ: may you share it? |
| 10:02.56 | TazzNZ | remote side ? |
| 10:03.02 | *** join/#asterisk tparcina (~tomo@212.92.192.54) |
| 10:03.04 | bulkorok | jssip ftw ;-) |
| 10:03.47 | DelphiWorld | TazzNZ: the remote pstn gw or phone dont need srtp |
| 10:03.50 | DelphiWorld | slaps bulkorok around a bit with a large trout |
| 10:05.16 | TazzNZ | your sip.conf needs to have srtp |
| 10:05.23 | TazzNZ | for the webrtc user |
| 10:05.55 | DelphiWorld | TazzNZ: yes this is sure:P |
| 10:05.56 | DelphiWorld | TazzNZ: can you share please? |
| 10:06.31 | TazzNZ | I can't share that one sorry |
| 10:06.44 | DelphiWorld | okay |
| 10:07.50 | TazzNZ | oh - now I remember - it was sipml5 |
| 10:08.05 | DelphiWorld | TazzNZ: sipml5 is complex and no much examples |
| 10:08.14 | TazzNZ | there is a live example ?!? |
| 10:08.25 | TazzNZ | http://sipml5.org/call.htm?svn=224 |
| 10:08.32 | DelphiWorld | TazzNZ: i'm not looking for a full webphone, but only for a click to call |
| 10:08.34 | TazzNZ | how hard is that |
| 10:09.35 | DelphiWorld | TazzNZ: no need a full phone, just C2C |
| 10:10.27 | TazzNZ | that is what I would use, anyways - off to bed |
| 10:10.30 | TazzNZ | later all |
| 10:10.56 | DelphiWorld | later TazzNZ |
| 10:11.40 | *** join/#asterisk jhlavacek (~jirka@80.82.225.246) |
| 10:12.26 | *** join/#asterisk workingcats (~workingca@212.122.48.77) |
| 10:12.52 | *** join/#asterisk calum_ (~calum_@cpc67428-harg5-2-0-cust142.7-1.cable.virginm.net) |
| 10:21.59 | *** join/#asterisk Greek-Boy (~Greek-Boy@41.76.92.218) |
| 10:26.27 | DelphiWorld | slap Greek-Boy with G.722.1 |
| 10:28.27 | *** join/#asterisk jepperl (~jepperl@213.150.61.162) |
| 10:29.36 | jepperl | Hello there :) Does anyone have any information available on when the Asterisk WIKI site will be back up? |
| 10:31.12 | D-Boy | Hi, https://wiki.asterisk.org/wiki/ works for you ? |
| 10:31.33 | eirirs | slaps DelphiWorld with ulaw. |
| 10:31.34 | jepperl | No, i get 503 Service Unavailavle |
| 10:31.50 | DelphiWorld | slap eirirs with alaw |
| 10:31.54 | D-Boy | jepperl: me too, so i guess that's a server side issue not mine ^^ |
| 10:31.58 | eirirs | damn american law |
| 10:32.07 | jepperl | Yea, i figured :) |
| 10:32.34 | jepperl | Can someone help me out with the format and return values of invoking an AGI script through the AMI ? |
| 10:32.54 | jepperl | Was going to look it up on the WIKI, but, yea :p |
| 10:33.53 | *** join/#asterisk jhlavacek (~jirka@80.82.225.246) |
| 10:34.14 | r00f | is http://www.voip-info.org considered outdated? |
| 10:35.09 | jepperl | r00f: i consider it somewhat uncomplete, but dont know what others think |
| 10:35.24 | coppice | r00f: its a bit like the way back machine |
| 10:36.11 | r00f | damn. have always looked there, so bad of me ( |
| 10:36.18 | coppice | the problem with sites like voip-info is nothing is date stamped so you can't tell what is old and what is new. that usually leads to the place evebtually being abandoned |
| 10:36.26 | D-Boy | jepperl: trying to browse the wiki using the wayback machine ==> https://web.archive.org/web/20140408221652/https://wiki.asterisk.org/wiki/display/AST/Getting+Started <== it's messy but readable |
| 10:36.41 | jepperl | r00f: You've been missing out man.. WIKI for president |
| 10:36.54 | *** join/#asterisk timahvo1 (~rogue@197.237.131.169) |
| 10:37.05 | r00f | but it never gave me bad advises. i guess i was lucky ) |
| 10:37.07 | r00f | thanks |
| 10:37.10 | jepperl | D-Boy: Thanks mate.. Pretty messy yea xD Ill play around with it |
| 10:37.23 | *** join/#asterisk timahvo1 (~rogue@197.237.131.169) |
| 10:44.22 | D-Boy | jepperl: np |
| 11:02.03 | *** join/#asterisk workingcats (~workingca@212.122.48.77) |
| 11:02.49 | *** join/#asterisk Zogot (~Adium@D4B2620B.static.ziggozakelijk.nl) |
| 11:13.56 | Zogot | any people with some experience with aastra phones? |
| 11:14.08 | Zogot | specially in relation to provisioning |
| 11:22.05 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
| 11:27.04 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 11:32.17 | *** join/#asterisk catphish (~catphish@unaffiliated/catphish) |
| 11:34.23 | Zogot | o found it, knew there was a better way |
| 11:34.27 | catphish | does asterisk have anything like fail2ban to blacklist IPs who made repeated failed registrations or invites? |
| 11:34.51 | Zogot | catphish: i dont think so, doesn't seem like something asterisk itself would control |
| 11:35.19 | r00f | why not use fail2ban? it has policies for asterisk oob |
| 11:35.23 | catphish | well, there's a couple of places it could happen, the simplest way would be the fail2ban approach, monitor the log and add iptables rules |
| 11:35.35 | catphish | r00f: oh, i didn't know that, i will look into it :) |
| 11:35.42 | catphish | thanks |
| 11:36.02 | r00f | later versions at least. i remember old ones, they didn't |
| 11:36.05 | Zogot | r00f: asterisk 11? |
| 11:36.10 | Zogot | o cool |
| 11:36.25 | catphish | i'm using quite an aging asterisk, need to get me upgraded at some point |
| 11:36.26 | r00f | i'm talking about fail2ban, not asterisk |
| 11:36.46 | catphish | but yeah, hopefully i can configure it to point to the asterisk logs :) |
| 11:36.48 | r00f | you just need to change the log format if it differs |
| 11:37.19 | r00f | but - yes, i'm using it with asterisk 11 and the only thing i've changed was my email for notifications |
| 11:42.29 | woopstar | I dont get this. Im using SipML5 with websocket secure to Asterisk. Connected fine. Encryption is enabled in sip.conf and for the extension. But when i try to call something, i get this in the logs: "Rejecting secure audio stream without encryption details: audio 51340 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126" |
| 11:44.37 | Greek-Boy | i've tried everything but can't get line appearance to turn from amber to red using blf fucntion on my SPA942 :-( |
| 11:52.21 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 11:53.08 | *** join/#asterisk petris (~petris@2607:5300:60:5475:a9d7:583:99b:a901) |
| 12:12.39 | *** join/#asterisk jhlavacek (~jirka@80.82.225.246) |
| 12:16.33 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:26.44 | *** join/#asterisk alami (~ialami@unaffiliated/alami) |
| 12:29.19 | alami | hello, how can i fix this http://pastie.org/9222834 |
| 12:31.11 | *** join/#asterisk jhlavacek (~jirka@80.82.225.246) |
| 12:31.25 | *** join/#asterisk jkroon (~jkroon@kantoor.wdns.uls.co.za) |
| 12:32.02 | alami | can i change the codec type? |
| 12:32.11 | jkroon | hi guys, assuming I have a queue with callers waiting (say "myqueue"), and a channel in my dialplan, is there any way that I can associate that channel with the next caller in the queue? |
| 12:32.51 | WIMPy | Redirect the call? |
| 12:34.30 | alami | WIMPy: is that for me? |
| 12:35.10 | WIMPy | no. You need to install the missing parts, or enable them. |
| 12:35.46 | alami | WIMPy: do you mean i have to install g723 ? |
| 12:36.06 | alami | or in sip.conf allow=ulaw allow=ulaw allow=gsm? |
| 12:36.18 | [TK]D-Fender | alami: Your call is in G.723 and is not a codec * supports except if you have a very expensive trancoder card. Choose another codec to use. |
| 12:36.51 | [TK]D-Fender | alami: You need to do "disallow=all" first, THEN do your "allow=" statments to specify which ones to allow. |
| 12:37.41 | *** join/#asterisk bmurt (~brendan@static-96-245-76-214.phlapa.fios.verizon.net) |
| 12:37.44 | file | wiki.asterisk.org update: I've opened a ticket with IT to get it back online. |
| 12:38.26 | jkroon | WIMPy, ok, so for that matter, how do I figure out what the next call in the queue is? Basically I want to provide an "extension" that users can dial to fetch the next call in the queue, if there are any. |
| 12:39.00 | WIMPy | I guess you have to keep the list yourself. |
| 12:39.36 | [TK]D-Fender | jkroon: one solution : You'd have to run a script to ID the caller in that position along with the channel of the waiting channel. You'd then SetVar the target channel into a variable on that caller's channel, and be calling a Local channel from your queue that will look at the caller's variable and do a Bridge |
| 12:41.51 | jkroon | [TK]D-Fender, ok, i'll re-read that - sounds rather crazy. |
| 12:42.42 | [TK]D-Fender | jkroon: Thinking you can just reserve agents for people is crazy.... The solution will clearly end up being somewhat hackish. |
| 12:48.44 | alami | [TK]D-Fender: thanks a solve it just by dowing allow=ulaw |
| 13:07.17 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 13:09.55 | Dovid | Dies anyone know where in the RFC it says that the ACK goes to the last IP in the RR? |
| 13:12.06 | *** join/#asterisk jhlavacek (~jirka@80.82.225.246) |
| 13:21.05 | bulkorok | Dovid: look at rfv3261 section 16.12.1 Examples |
| 13:21.14 | bulkorok | rfc3261 |
| 13:22.05 | bulkorok | it's not going to the last… record-route makes the uac sending packets to the top record-route entry… top to second and so on... |
| 13:23.38 | Dovid | in Asterisk I have set outboundproxy=OpenSipS. the issue is that the ACK from Asterisk1 is set to the IP of OpenSipS-B When it should be set to Asterisk2 so when when OpensIPs-B gets it, it does not pass it to Asterisk2. Asterisk2 then never gets the ACK and after 30 seconds hangsup up the call |
| 13:25.29 | *** join/#asterisk saint_ (~saint@c-50-166-85-78.hsd1.nj.comcast.net) |
| 13:26.16 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:28.44 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:307e:45b9:ac41:5681) |
| 13:30.22 | bulkorok | outboundproxy != record-route |
| 13:31.01 | Dovid | bulkorok: So it's OpenSipS that needs to add that RR ? |
| 13:31.10 | bulkorok | yeah |
| 13:31.46 | bulkorok | what do you have? asterisk1 <=> opensips <=> asterisk2 ?! |
| 13:32.36 | Dovid | Asterisk1 (10.10.10.1) -> OpenSipS1 (10.10.10.2) -> OpenSipS2 (10.100.100.2) -> Asterisk2 (10.100.100.1). The peer in Asterisk1 (where it is sending the call to) is bob@10.100.100.2. It does not know about the Asterisk server (10.100.100.1). Also the peer is set up with outboundproxy=10.10.10.2. On Asterisk2 it's outboundproxy is 10.100.100.2 |
| 13:34.07 | bulkorok | sounds very(!) complicated |
| 13:34.25 | bulkorok | you should try it with rr-module in opensips and define a proper routing in them |
| 13:34.29 | Dovid | bulkorok: yes but should still work. |
| 13:34.39 | bulkorok | it doesn't ;-) |
| 13:34.50 | Dovid | i have it working in another set up and i dont know why |
| 13:34.59 | bulkorok | ok... |
| 13:35.18 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-dlwvuvlwsauaekvp) |
| 13:35.19 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 13:35.19 | bulkorok | maybe you can compare the sip-headers and can spot the differences |
| 13:37.01 | Dovid | i am trying to do that. in the wone that works where i see it sending the ack to the last destination I see: under all of the RecordRoutes: m: <SIP:100.100.100.100:5060;transporrt=UDP |
| 13:37.06 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-gqwevdbmvfipxube) |
| 13:37.06 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:37.29 | Dovid | Wireshark says that is the contact URI. but i cant see in the RFC where its supposed to use that that |
| 13:38.50 | *** join/#asterisk jhlavacek (~jirka@80.82.225.246) |
| 13:44.14 | *** join/#asterisk mattcen (~mattcen@c110-22-201-130.sunsh4.vic.optusnet.com.au) |
| 13:50.44 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-xngtmcjrcsjhsrpe) |
| 13:51.12 | *** join/#asterisk yaaago (~kresp0@gateway/tor-sasl/kresp0) |
| 13:53.37 | *** part/#asterisk bulkorok (~Adium@85.183.61.47) |
| 13:59.38 | *** join/#asterisk solmsted (~solmsted@pool-71-176-247-26.rcmdva.fios.verizon.net) |
| 14:00.26 | *** join/#asterisk jhlavacek (~jirka@80.82.225.246) |
| 14:04.57 | *** join/#asterisk DougsTech (~DougsTech@c-98-230-105-210.hsd1.al.comcast.net) |
| 14:07.38 | *** join/#asterisk aness (~aness@cm-84.215.80.229.getinternet.no) |
| 14:09.33 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:09.33 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:23.27 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 14:27.36 | *** join/#asterisk Draecos (~Draecos@124-148-114-147.dyn.iinet.net.au) |
| 14:30.39 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 14:31.14 | *** join/#asterisk mattcen (~mattcen@c110-22-201-130.sunsh4.vic.optusnet.com.au) |
| 14:32.55 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:32.55 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:33.26 | *** join/#asterisk `mc (~mc@c-71-235-50-162.hsd1.ct.comcast.net) |
| 14:33.28 | `mc | hello |
| 14:43.11 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 14:47.44 | *** join/#asterisk jhlavacek (~jirka@80.82.225.246) |
| 14:50.03 | *** join/#asterisk ttyS3 (~o@gateway/tor-sasl/omlib) |
| 15:00.40 | *** join/#asterisk Draecos (~Draecos@124-148-114-147.dyn.iinet.net.au) |
| 15:07.27 | *** join/#asterisk navaismo (~navaismo@200-52-45-221.dynamic.axtel.net) |
| 15:18.04 | *** join/#asterisk jhlavacek (~jirka@80.82.225.246) |
| 15:19.11 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:21.42 | *** join/#asterisk spditner (~simon@206-248-183-93.dsl.teksavvy.com) |
| 15:28.40 | marceloamorim | guys, how can I do this, I call A to B and this call set some variables, then I need to call B for C, how can I do B for C receive those variables that I set before? |
| 15:30.02 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 15:30.03 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 15:31.25 | navaismo | using global vars or astdb |
| 15:36.49 | *** join/#asterisk jaflong (5bec7504@gateway/web/freenode/ip.91.236.117.4) |
| 15:39.50 | jaflong | Hi all, in the dialplan is there a way to get the rtp ports |
| 15:41.35 | [TK]D-Fender | Not directly. |
| 15:41.48 | [TK]D-Fender | You'd have to check with CLI/AMI, etc to get that |
| 15:43.02 | marceloamorim | I´m trying using the global, but didn't work yet, I'll keep trying ty navaismo |
| 15:51.40 | [TK]D-Fender | drilling CLI I can get the SIP channel ID, but a dump doesn't list the port, and I see no CLI command to get that.... that leaves AMI at best. It might not be passed anywhere usable |
| 15:53.12 | jaflong | Dial app manages to forward the ports somehow |
| 15:54.29 | Qwell | it does no such thing |
| 15:55.42 | [TK]D-Fender | "forward the ports" doesn't really mean anything.... |
| 15:58.35 | jaflong | I mean it pickups up the sdp m=audio value from somewhere to send to the dial destination |
| 15:59.29 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 15:59.30 | [TK]D-Fender | jaflong: * has access the the raw structures. There simply isn't a variable or function anywhere, or any place I can see it dumpable |
| 16:01.12 | jaflong | how about the whole sdp and I parse for what I need. is the sdp saved anywhere |
| 16:02.24 | file | you can use the CHANNEL dialplan function on SIP channels via ${CHANNEL(rtpdest,audio)} and ${CHANNEL(rtpsource,audio)} |
| 16:04.12 | [TK]D-Fender | file: Oh, will that give the port as well? |
| 16:05.15 | file | hum |
| 16:05.22 | file | yes |
| 16:05.25 | file | <IP>:<port> |
| 16:07.00 | jaflong | will this work for video as well |
| 16:08.29 | file | if you specify video instead of audio, yes |
| 16:09.14 | jaflong | this is only available for 1.6 and over? |
| 16:10.02 | file | I only remember information for supported versions. |
| 16:10.47 | jaflong | ok thanks for the help |
| 16:22.46 | *** join/#asterisk [[thufir]] (~thufir@S0106c8fb2677e8ca.vs.shawcable.net) |
| 16:26.58 | *** join/#asterisk mattcen (~mattcen@c110-22-201-130.sunsh4.vic.optusnet.com.au) |
| 16:27.23 | *** join/#asterisk bkruse (~Adium@64.89.97.127) |
| 16:29.39 | *** join/#asterisk Zogot (~Adium@90-145-116-55.bbserv.nl) |
| 16:33.51 | *** join/#asterisk dumby (~dumby@204.246.140.162) |
| 16:37.23 | *** join/#asterisk Qwell (north@pdpc/sponsor/digium/Qwell) |
| 16:37.23 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 16:49.27 | *** join/#asterisk jpoz (~jpoz@107-1-105-37-ip-static.hfc.comcastbusiness.net) |
| 16:54.25 | *** join/#asterisk jpoz (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
| 16:55.45 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 17:10.36 | *** join/#asterisk zerick (~eocrospom@190.118.43.113) |
| 17:12.37 | *** join/#asterisk marvinmartian (~marvinmar@64.237.52.137) |
| 17:13.55 | marvinmartian | How can you get voicemail to emails to send to multiple addresses? Would you just add a comma and another email to this line in voicemail.conf? 54 => 5678,John Henry,jhenry@gmail.com |
| 17:14.09 | [TK]D-Fender | no. You'd have to do this in your MTA |
| 17:17.27 | marvinmartian | Thank you. I'm not sure if this will work for postfix. I'm not familiar with it. What do you think? http://www.thirdlane.com/forum/sending-a-voicemail-to-multiple-e-mails |
| 17:18.44 | leifmadsen | email a mailing list? |
| 17:18.49 | leifmadsen | that's what it sounds like :) |
| 17:22.03 | marvinmartian | thank you |
| 17:30.09 | *** join/#asterisk sarthor (~sarthor@unaffiliated/sarthor) |
| 17:30.30 | sarthor | HI, How to check that my ISP is not blocking ports 5060-5070 and from 10000 to 20000, I have Asterisk installed, and Extensions can easily connect from local network, but not from Internet, |
| 17:32.54 | [TK]D-Fender | try to connect. |
| 17:32.58 | [TK]D-Fender | That is your test |
| 17:33.32 | [TK]D-Fender | and confirm what you have actually set up and describe the full networking path between the two |
| 17:56.01 | *** join/#asterisk CeBe (~CeBe@brsg-4dbbda48.pool.mediaWays.net) |
| 17:58.40 | marceloamorim | guys, I use the Set(GLOBAL(ORIG=${CALLERID(DNID))}); then I tried to transfer to number 200 and this number do Set(test1=${IMPORT(SIP/guilherme,ORIG)); but the NoOp to this test1 is null |
| 17:59.24 | marceloamorim | dumping the channel: Dumping Info For Channel: SIP/guilherme-000007e7: and |
| 17:59.25 | marceloamorim | Variables GLOBAL(ORIG=113) |
| 18:00.18 | navaismo | ${ORIG} |
| 18:00.30 | marceloamorim | ow |
| 18:01.21 | marceloamorim | cool |
| 18:01.27 | marceloamorim | it works |
| 18:01.33 | marceloamorim | thx navaismo =) |
| 18:01.39 | navaismo | np |
| 18:02.24 | marceloamorim | Now I need to think how can I do this automatic, like I set the SIP/guilherme |
| 18:02.35 | marceloamorim | but I need to do this for any channel |
| 18:03.04 | navaismo | what are you trying to do? |
| 18:03.15 | marceloamorim | I'll post my progress here =) |
| 18:03.35 | marceloamorim | well, I want to do like a research with the clients |
| 18:04.59 | marceloamorim | I need to do this after the call, I'll transfer the call for some number that do this research with the client, but I need to pass the callerid, the phone that receive this call and the number that the client choose |
| 18:06.34 | marceloamorim | I do this like context ramais -> 200 => { set those variables than goto to URA and at the URA I use the shell script to get the results |
| 18:15.26 | leifmadsen | marceloamorim: you're using it wrong |
| 18:15.41 | *** join/#asterisk calum_ (~calum_@host109-154-13-46.range109-154.btcentralplus.com) |
| 18:15.41 | leifmadsen | Set(GLOBAL(ORIG)=${CALLERID(dnid)}) |
| 18:18.04 | marceloamorim | but the variables isn´t work until now, now the only problem is the IMPORT(channel) right channel |
| 18:18.19 | marceloamorim | I'll try this ) at the end of variable |
| 18:23.14 | navaismo | } |
| 18:24.47 | *** join/#asterisk jwr_ (~quassel@205.196.167.54) |
| 18:27.40 | *** join/#asterisk jql (~awinters@74.123.82.143) |
| 18:31.44 | leifmadsen | just saying your syntax was wrong |
| 18:32.08 | *** join/#asterisk CeBe (~CeBe@brsg-4dbbda48.pool.mediaWays.net) |
| 18:32.14 | navaismo | that too |
| 18:33.43 | *** join/#asterisk u0m3 (~u0m3@92.80.122.72) |
| 18:40.59 | *** join/#asterisk u0m3 (~u0m3@92.80.122.72) |
| 18:52.43 | *** join/#asterisk panic788 (62ac15b1@gateway/web/freenode/ip.98.172.21.177) |
| 18:52.53 | panic788 | hello... |
| 18:54.53 | panic788 | my full file is well, filling up my disk |
| 18:55.02 | panic788 | any easy way to remedy that? |
| 18:55.35 | [TK]D-Fender | man rm |
| 18:55.40 | navaismo | file stop filling his disk |
| 18:56.11 | panic788 | haha, its a legacy thing at my work that was just thrown in my lap |
| 18:56.37 | panic788 | i didnt know if I could just rm full, and not blow something important away |
| 18:56.58 | navaismo | or using dd, qwel has a very nice method to use the mouse for dd your disk |
| 18:59.09 | file | moo |
| 19:00.10 | panic788 | dd? qwel? |
| 19:00.17 | panic788 | lost me there |
| 19:00.49 | navaismo | just thinking-writing safely discard my brain-farts |
| 19:01.14 | panic788 | roger |
| 19:04.11 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 19:09.32 | panic788 | navaismo, is /var/log/asterisk/full just a verbose copy of the cli that has been running since it was setup? |
| 19:10.11 | navaismo | if was configured since the beginning of that machine, yes |
| 19:10.19 | navaismo | its the log of asterisk |
| 19:10.44 | panic788 | im making lots of assumptions, for better or probably worse |
| 19:11.07 | panic788 | inhereted system that I really know nothing about |
| 19:11.10 | navaismo | you can compress it |
| 19:11.38 | navaismo | a good practice is to logrotate it every month or week depending on your call load |
| 19:11.51 | panic788 | what i have read thus far leads me to think it wasnt properly setup to rotate out of the box |
| 19:12.08 | panic788 | so this is probably logs going back 6 years |
| 19:12.20 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 19:12.22 | navaismo | amm as far I know you need to setup is not out of the box at least not for my plain asterisk installations |
| 19:12.32 | navaismo | is this a plain asterisk or canned distro? |
| 19:13.03 | panic788 | yes? |
| 19:13.19 | panic788 | i know it has freepbx on top of it |
| 19:13.34 | navaismo | ok, not plain asterisk |
| 19:13.59 | navaismo | what is the size? |
| 19:14.16 | panic788 | size of the file? |
| 19:14.19 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 19:14.20 | navaismo | of you full log |
| 19:14.28 | panic788 | 65gigs |
| 19:14.32 | navaismo | awesome |
| 19:14.39 | panic788 | haha |
| 19:14.45 | navaismo | if you can rm that log |
| 19:14.53 | panic788 | im assuming sarcasm |
| 19:15.02 | *** join/#asterisk woopstar (~woopstar@unaffiliated/woopstar) |
| 19:15.07 | navaismo | about the rm? |
| 19:15.11 | woopstar | Is Asterisk 12.2.0 production ready? |
| 19:15.17 | panic788 | no, about the size of the log |
| 19:15.24 | navaismo | nope |
| 19:15.32 | Chainsaw | No, but the 11 branch is. |
| 19:15.33 | [TK]D-Fender | woopstar: It's about as reliable as the next thing, just not an LTS branch. |
| 19:15.39 | navaismo | woopstar, is not LTS but you can use it |
| 19:15.43 | navaismo | too late |
| 19:16.04 | woopstar | So.. it should be more or less production ready, but hot fixes can come |
| 19:16.06 | woopstar | right? |
| 19:16.06 | [TK]D-Fender | woopstar: If you expect to let you box sit on a brnach for a long while, go with 11 for now. |
| 19:16.39 | navaismo | panic788, i really recommend to delete the log. add to the logrotate and start over. Or move it to the "backup disk" |
| 19:16.50 | Chainsaw | "Production" is a subjective word in the end, but in the meaning of "let it set for months/years and be okay", you want 11. |
| 19:16.56 | [TK]D-Fender | woopstar: 11 & 12 are both functionally stable. It's a question of if you are looking to build for the long-term, vs being ready to upgrade 12 when it EOL's sooner |
| 19:16.57 | Chainsaw | 12 isn't LTS |
| 19:17.05 | *** join/#asterisk CeBe (~CeBe@brsg-4dbbda48.pool.mediaWays.net) |
| 19:17.11 | woopstar | Thing is... After Chrome was updated to 35+ they removed SDES and now rely on DTLS-SRTP. And I cannot get SipML5 to work with 11.9.0 at all means |
| 19:17.34 | Chainsaw | set? sit. |
| 19:17.34 | Chainsaw | Sorry. |
| 19:17.38 | navaismo | hehe |
| 19:17.53 | panic788 | navaismo, that was the direction I was leaning towards, but I didn't want to nuke something that I know little to nothing about that is in use for the small office I am in |
| 19:18.24 | woopstar | So I was like.. Ok.. maybe I can get it to work with 12.2.0.. but wouldn't waste time if it was too unstable to production... |
| 19:18.37 | panic788 | if it is just a log of extensionxyz is on, and now there is a call coming in, so I am ringing this group or that, and so on, then I won't worry about it |
| 19:19.30 | panic788 | then I can go about setting up the logrotate.d part up |
| 19:20.26 | navaismo | woopstar, i love this tweet--->https://twitter.com/ibc_tw/status/441346327771746304 |
| 19:20.50 | woopstar | navaismo: going to read it now.. :) |
| 19:21.28 | panic788 | it is just a log like that, right? |
| 19:21.34 | panic788 | before i pull the trigger |
| 19:22.04 | navaismo | panic788, full log contain the verbose of the asterisk cli and if is enable also the debug of asterisk so it is usefull to troubleshoot issues but if you dont want to checkout the log in the nearest future remove it if not backup |
| 19:23.05 | woopstar | navaismo: so it seems... it's all fucked... as usual :( |
| 19:23.13 | panic788 | gotcha...space is cheap, i can back it up if I need to look later, and then nuke it so it goes for now |
| 19:23.57 | navaismo | woopstar, since webrtc is not a standard not have any RFC and any developer are using it as he wants then is working as expected |
| 19:24.37 | woopstar | but thing is.. i cant get it to connect... it's when doing calls i keep getting the problems |
| 19:24.55 | woopstar | but seems related to the DTLS/SDES thingy |
| 19:24.58 | navaismo | yep |
| 19:25.11 | navaismo | is becasue that, your best and fast solution roll back chrome 34 |
| 19:25.25 | navaismo | or enable dtls in your asterisk or use the webrtc2sip gateway |
| 19:25.36 | woopstar | but i did enable dtls in my asterisk :( |
| 19:25.57 | woopstar | webrtc2sip gateway is soooo broken :( |
| 19:25.59 | woopstar | and unstable |
| 19:27.35 | navaismo | ^depends a lot on the setup |
| 19:27.42 | woopstar | I added this to the general section in my sip conf : http://pastebin.com/dAy1VHiG (well ofc I configured it) |
| 19:27.56 | navaismo | but then again you are using a HYPE technology not a standard one |
| 19:28.21 | woopstar | I kinda like to break off the webrtc2sip and just let SipML connect to Asterisk |
| 19:28.53 | navaismo | https://issues.asterisk.org/jira/browse/ASTERISK-22961 |
| 19:29.50 | woopstar | So still fucked? |
| 19:30.10 | woopstar | thing is.. audio was broken for me with webrtc2sip after Chrome 35 |
| 19:30.11 | woopstar | :/ |
| 19:31.52 | navaismo | whisper with rape accent: roll back to chrome 34 |
| 19:32.05 | woopstar | :( |
| 19:32.17 | woopstar | hates life |
| 19:32.46 | navaismo | ~life |
| 19:32.46 | infobot | 42 (duh.) |
| 19:32.57 | woopstar | its all 42... |
| 19:33.06 | navaismo | i prefer 69 |
| 19:34.41 | navaismo | https://groups.google.com/d/msg/doubango/iQ_8rgpbm0k/uVGgXKnW8GUJ |
| 19:35.01 | *** join/#asterisk jasonwert (~w3rt@71.89.137.28) |
| 19:35.58 | woopstar | yeah, read that eailer |
| 19:36.25 | navaismo | https://groups.google.com/forum/#!topic/doubango/SkwpTApyi3g |
| 19:38.39 | woopstar | guess my best bet is the webrtc2sip gateway |
| 19:38.52 | navaismo | whats the issue rolling back chrome? |
| 19:38.55 | woopstar | until asterisk and webrtc (chrome) is aligned.. |
| 19:39.14 | woopstar | it currently running for users where the option to rollback is even more a pain in the ass.. |
| 19:39.16 | woopstar | :( |
| 19:39.51 | navaismo | hmmm http://www.myfacewhen.net/uploads/2058-thinking.jpg |
| 19:40.05 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 19:40.44 | woopstar | But I think the matters would be resolved within weeks |
| 19:40.51 | woopstar | due to developers and such |
| 19:43.07 | *** join/#asterisk kleszcz (tick@ip-178-216-200-29.e24cloud.com) |
| 19:46.29 | navaismo | i guess you are-->http://forums.digium.com/viewtopic.php?f=1&t=90428&sid=2d874b65e3052eeb070ee4ba8116adb6 |
| 19:47.46 | woopstar | guess i need to follow that thread too :D |
| 19:47.48 | marceloamorim | there is a variable when you show the dumpchan, the name is INFO: Name=SIP/guilherme-000009ee, Type=SIP, I wish to use those too variables, Name and Type, is that possible? |
| 19:48.35 | navaismo | woopstar, follow the JIRA page |
| 19:48.44 | woopstar | yeah |
| 19:59.39 | *** join/#asterisk Zogot (~Adium@90-145-116-55.bbserv.nl) |
| 20:04.17 | navaismo | marceloamorim, you can use CHANNEL(name) & CHANNEL(channeltype) |
| 20:04.25 | navaismo | more info core show function CHANNEL |
| 20:12.34 | *** join/#asterisk darkdrgn2k (~darkdrgn2@209.90.253.66) |
| 20:13.31 | darkdrgn2k | is it possible to add a video stream mid way... IE: i have an FXS thats hooked up to a door panel, i have a camera on it as well thats wired to a capature card. i want the FXS to call an asterisk box, the asterisk box to add the video stream and send it to whoever it supposed to call. |
| 20:18.16 | [TK]D-Fender | There is no combining from multiple sources |
| 20:21.06 | darkdrgn2k | so cant do audio from a and video from b :) |
| 20:21.06 | darkdrgn2k | :( |
| 20:23.45 | `mc | anyone here available and interested in a contracting opportunity? |
| 20:24.05 | `mc | Need an experienced asterisk developer |
| 20:25.59 | `mc | if you are, please /msg me |
| 20:27.53 | *** join/#asterisk Zogot (~Adium@90-145-116-55.bbserv.nl) |
| 20:29.54 | marceloamorim | it works guys, my file has callerid from the first channel, callerid of the second phone and then the number that the client choose |
| 20:30.53 | marceloamorim | I just need to clear the global var of the end of this call, otherwise the research isn't validate |
| 20:31.22 | marceloamorim | other problem is the multiples call at the same time =( |
| 20:31.26 | marceloamorim | crap |
| 20:31.37 | marceloamorim | I didn't think about that =( |
| 20:32.08 | navaismo | try switching to astdb, create the variables appending the channel UID then at the end of the call delete those vars from the astdb |
| 20:35.05 | marceloamorim | I'll try to change |
| 20:35.20 | marceloamorim | ty for this tip |
| 20:56.48 | *** join/#asterisk jpoz (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
| 21:05.56 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 21:05.56 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 21:19.03 | *** join/#asterisk Cynagen (~cynagen@ip70-190-135-8.ph.ph.cox.net) |
| 21:20.10 | *** join/#asterisk calum_ (~calum_@cpc67428-harg5-2-0-cust142.7-1.cable.virginm.net) |
| 21:21.35 | marceloamorim | navaismo: are you there? |
| 21:26.06 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 21:27.21 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 21:32.03 | *** join/#asterisk jpoz (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
| 21:32.55 | navaismo | yep |
| 21:34.03 | marceloamorim | I tried this astdb, now I'm saving this calls on /Pesquisa/calls/id/src : callerid |
| 21:34.16 | marceloamorim | when I use database showkey src |
| 21:34.42 | marceloamorim | the cli shows me the callerid and always save on diferent ID for everycall, everything perfect |
| 21:35.02 | marceloamorim | but how can I use this information? |
| 21:36.07 | marceloamorim | maybe the best question is, how can I get this on another channel |
| 21:36.24 | *** join/#asterisk wolrah (~wolrah@24.239.210.140) |
| 21:37.31 | navaismo | hmmm how do you associate the channels do you use a macro or something, because passing arguments is the only way(i can imagine right now) to retrieve that |
| 21:38.19 | *** join/#asterisk Alex_Bkash (7c06ecfc@gateway/web/freenode/ip.124.6.236.252) |
| 21:38.40 | marceloamorim | Set(uniqIdpesquisa=${CUT(UNIQUEID,.,1)}); |
| 21:38.42 | marceloamorim | Set(DB(Pesquisa/calls/${uniqIdpesquisa}/src)=${CALLERID(num)}); |
| 21:38.42 | marceloamorim | Set(orig=${DB(Pesquisa/calls/${uniqIdpesquisa}/src)}); |
| 21:39.24 | marceloamorim | this way I always save the callerid per UNIQUEID |
| 21:40.27 | navaismo | are you terminating the commands with an empty comment? |
| 21:40.48 | navaismo | or its a behavior coming from php ; |
| 21:40.53 | marceloamorim | ael |
| 21:41.03 | navaismo | oh i see |
| 21:41.04 | marceloamorim | so terminando with ; |
| 21:41.09 | navaismo | yep |
| 21:41.16 | marceloamorim | terminating with this ; |
| 21:41.41 | marceloamorim | its perfect when I use on cli: database showkey src |
| 21:41.55 | marceloamorim | it gives me the correct callerid that I need it |
| 21:42.40 | marceloamorim | but I'm stuck on how I can get this correct callerid on another channel |
| 21:42.44 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 21:42.57 | navaismo | how do you open the other channel? |
| 21:43.55 | marceloamorim | I open with a transfer calling for a number that I set a goto for an URA |
| 21:44.35 | marceloamorim | so I transfer to number 200 and on the number 200 I use goto URA|s|test; |
| 21:45.23 | marceloamorim | then on the URA, I have numbers calling for a shell script |
| 21:46.15 | navaismo | hmmm |
| 21:46.34 | marceloamorim | if you don't get my english, I can pastebin for you |
| 21:46.50 | navaismo | i understand your english but paste it |
| 21:46.57 | marceloamorim | ok |
| 21:48.54 | marceloamorim | http://pastebin.com/QAR7Xchu |
| 21:51.13 | marceloamorim | I did't paste all from_clients2 because is huge |
| 21:53.00 | marceloamorim | the 200 => is only for goto atendimento_automatico3, I did't test the DBGet yet |
| 22:02.08 | marceloamorim | btw, the atendimento_automatico3 is using the variables that I use with GLOBAL var |
| 22:02.21 | marceloamorim | used* |
| 22:03.47 | marceloamorim | oh boy, I got go =((( |
| 22:04.55 | marceloamorim | tomorrow I'm back to this problem, so I see if you or anybody answered something |
| 22:07.23 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 22:12.01 | navaismo | sorry im on a ticket job, so this monkekey needs to eat |
| 22:13.25 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 22:14.19 | *** join/#asterisk dumby (~dumby@204.246.140.162) |
| 22:21.01 | *** join/#asterisk dumby_PC (~dumby@204.246.140.162) |
| 22:23.30 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:24.23 | *** join/#asterisk Gugge (gugge@kriminel.dk) |
| 22:29.02 | *** join/#asterisk petris (~petris@2607:5300:60:5475:a9d7:583:99b:a901) |
| 22:30.30 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 22:40.30 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 23:09.38 | *** part/#asterisk marvinmartian (~marvinmar@64.237.52.137) |
| 23:20.13 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-bzajyqahlghxdgrx) |
| 23:20.13 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 23:32.00 | *** join/#asterisk zerohalo (~zerohalo@2601:6:f80:224:580c:f496:acfd:2574) |
| 23:51.25 | *** join/#asterisk theron (~theron@ip66-43-220-25.static.ishsi.com) |
| 23:57.54 | *** join/#asterisk acassio (~acassio@177.159.53.132) |
