| 00:24.50 | *** join/#asterisk marlinc (~marlinc@bouncer.cvo.technology) |
| 00:26.05 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 00:35.17 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 00:40.02 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 00:52.38 | *** join/#asterisk X-Rob (sid14615@gateway/web/irccloud.com/x-urllquyhbgwbhfiy) |
| 00:55.17 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-bmmwjfeteuxswngi) |
| 00:59.57 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 02:20.30 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 02:29.15 | *** join/#asterisk Oatmeal (~Suzeanne@75-103-145-152.ccrtc.com) |
| 02:30.35 | *** join/#asterisk axisys (~axisys@unaffiliated/axisys) |
| 02:47.09 | *** join/#asterisk Oatmeal (~Suzeanne@75-103-145-152.ccrtc.com) |
| 02:48.37 | *** join/#asterisk UncleKiwi (~UncleKiwi@unaffiliated/unclekiwi) |
| 02:57.57 | *** join/#asterisk Oatmeal (~Suzeanne@75-103-145-152.ccrtc.com) |
| 03:10.47 | *** join/#asterisk Oatmeal (~Suzeanne@75-103-145-152.ccrtc.com) |
| 03:13.57 | *** join/#asterisk vader- (~Adium@pool-71-175-67-97.phlapa.fios.verizon.net) |
| 03:15.50 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 04:30.25 | *** join/#asterisk robink_ (~quassel@unaffilated/robink) |
| 05:25.38 | *** join/#asterisk robmal (r@wporzo.pl) |
| 05:42.50 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 06:22.20 | *** join/#asterisk ganbold (~ganbold@173.244.215.173) |
| 06:27.48 | *** join/#asterisk EOIP (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 06:35.47 | *** join/#asterisk vader- (~Adium@pool-71-175-67-97.phlapa.fios.verizon.net) |
| 06:47.02 | *** join/#asterisk cesurasean (~cesurasea@c-68-35-152-116.hsd1.al.comcast.net) |
| 06:47.06 | cesurasean | anyone around? |
| 06:47.26 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 06:57.15 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 07:10.45 | *** join/#asterisk theron (~theron@173-28-183-190.client.mchsi.com) |
| 07:18.07 | *** join/#asterisk areski (~areski@80.174.128.118.dyn.user.ono.com) |
| 07:20.31 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 07:46.30 | *** join/#asterisk evil_gordita (robert@ip70-188-63-173.rn.hr.cox.net) |
| 07:49.00 | *** join/#asterisk pchero_work (~pchero@109.70.54.56) |
| 07:52.03 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 07:52.24 | *** join/#asterisk Ikikay (Ikikay@ADijon-156-1-28-162.w90-39.abo.wanadoo.fr) |
| 07:58.55 | cesurasean | how do i find out why voip.ms is giving me a busy signal? |
| 08:01.55 | *** join/#asterisk Zogot (~Adium@185.21.52.251) |
| 08:04.22 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 08:08.45 | *** join/#asterisk bluemerlin (~bluemerli@109.224.221.82) |
| 08:13.24 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 08:13.45 | *** join/#asterisk bulkorok (~Adium@89.245.151.228) |
| 08:14.00 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:16.51 | *** join/#asterisk samneo (~samneo@123.63.166.181) |
| 08:25.24 | samneo | hi, can anyone tell if we can set variable when using AMI originate command to initiate new call |
| 08:29.16 | *** join/#asterisk wonderworld (~ww@pD957A6A3.dip0.t-ipconnect.de) |
| 08:34.34 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 08:51.24 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 08:51.55 | *** join/#asterisk PolarbearIbiza (~Adrian@80.70.180.26) |
| 08:59.47 | PolarbearIbiza | Hey Dudes, i've been new to asterisk but in the last two days i got it working that i can make "internal" calls and that i can receive calls from outside - but i can't make calls to outside. I'm using sip from german telekom. If i make an outside call i got the following error: "Got SIP response 400 "Bad request" back from xxx.xxx.xxx.xxx:5060" and "SIP/t-online.de-0000000f is circuit-busy" does someone got a hint for me? thank you |
| 09:12.37 | *** join/#asterisk theron (~theron@173-28-183-190.client.mchsi.com) |
| 09:19.56 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 09:59.38 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 10:07.31 | *** join/#asterisk wonderworld (~ww@pD957AB83.dip0.t-ipconnect.de) |
| 10:27.30 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 10:34.52 | *** join/#asterisk ganbold (~ganbold@173.244.215.173) |
| 10:38.43 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 10:52.23 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 10:58.05 | *** join/#asterisk areski (~areski@106.Red-83-37-156.dynamicIP.rima-tde.net) |
| 10:59.54 | *** join/#asterisk lumasepa (~gestoip@193.145.124.30.local.ull.es) |
| 11:02.21 | *** join/#asterisk ThomasKeller (~Thomas@vmx.ethz.ch) |
| 11:12.38 | lumasepa | Hi, I'm looking for some help, I have a setup of two asterisk 13.6 in HA (Active - Pasive) via vrrp, so the active node in one interface has two ips the vip and a second one. I'm using the pjsip stack and asterisk is binded to the vip for SIP, but the problem comes with rtp, when asterisk sends an rtp package it uses the second ip rather than the vip (I think that asterisk is listening for rtp trafic in 0.0.0.0 because of that, it ends usin |
| 11:13.23 | lumasepa | If someone can help i will be very grateful |
| 11:13.35 | *** join/#asterisk theron (~theron@173-28-183-190.client.mchsi.com) |
| 11:25.56 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 11:31.18 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 11:54.54 | *** join/#asterisk puzzled (~patrick@2001:982:1097:1:70a7:41e6:9202:42c3) |
| 11:57.50 | *** join/#asterisk jonno11 (~Jon@cpc1-walt12-2-0-cust582.13-2.cable.virginm.net) |
| 11:59.03 | *** join/#asterisk vinrock (~vin@unaffiliated/vinrock) |
| 12:02.31 | *** join/#asterisk wonderworld (~ww@82.113.99.43) |
| 12:29.53 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:32.24 | [TK]D-Fender | samneo: yes |
| 12:32.32 | *** join/#asterisk CeBe (~CeBe@a81-14-224-229.net-htp.de) |
| 12:40.33 | *** join/#asterisk italorossi (~Adium@187.60.66.11) |
| 12:42.17 | *** join/#asterisk wonderworld (~ww@89.204.138.158) |
| 12:50.10 | *** part/#asterisk italorossi (~Adium@187.60.66.11) |
| 12:59.53 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 13:10.56 | *** join/#asterisk hirogen1 (b921d23e@gateway/web/freenode/ip.185.33.210.62) |
| 13:10.58 | hirogen1 | hi |
| 13:10.58 | hirogen1 | there |
| 13:11.27 | hirogen1 | a user when dialing out on her avaya x-one communicator software phone, its adding a 1 before the 9 therefore getting a constant engage tone |
| 13:11.35 | hirogen1 | on any number she tries to call |
| 13:11.37 | hirogen1 | any ideas |
| 13:12.41 | WIMPy | samneo: yes |
| 13:16.50 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-ziwaftlgjrtasruk) |
| 13:21.32 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 13:24.21 | *** join/#asterisk zapata (~zapata@2a02:b18:581:10:fdc9:772f:c04f:dd68) |
| 13:26.16 | [TK]D-Fender | hirogen1: Fix your dialplan or fix your softphone |
| 13:27.24 | hirogen1 | yah its effecting the one person |
| 13:27.26 | hirogen1 | dont know how |
| 13:27.32 | hirogen1 | will try a few things |
| 13:27.40 | cesurasean | for some reason my asterisk seems to start fine but doesnt open up ports |
| 13:27.46 | cesurasean | anyone have any idea how to check or fix this issue? |
| 13:28.24 | WIMPy | cesurasean: Do you load any modules that would do so? |
| 13:28.35 | cesurasean | no, just a standard debian install. |
| 13:28.37 | cesurasean | on jessie |
| 13:28.41 | cesurasean | their newest stable |
| 13:28.55 | WIMPy | How ancient is that? |
| 13:29.01 | cesurasean | not very |
| 13:29.08 | cesurasean | mainstream |
| 13:29.15 | cesurasean | actually |
| 13:29.17 | WIMPy | Which version? |
| 13:29.20 | cesurasean | Jessie |
| 13:29.23 | cesurasean | Debian Jessie |
| 13:29.24 | cesurasean | oh |
| 13:29.27 | cesurasean | version of asterisk? |
| 13:29.36 | WIMPy | yeah |
| 13:30.04 | cesurasean | how do i check? |
| 13:30.22 | WIMPy | It tells you when you start it. |
| 13:30.30 | cesurasean | not on debian it doesn't. |
| 13:31.34 | WIMPy | Is it even running? |
| 13:31.46 | WIMPy | Can you connect using 'rasterisk'? |
| 13:32.29 | cesurasean | yeah |
| 13:32.35 | cesurasean | it may just be a firewall issue, no? |
| 13:32.55 | WIMPy | If your problme description was wrong. |
| 13:33.12 | [TK]D-Fender | cesuraseanfor some reason my asterisk seems to start fine but doesnt open up ports <- * does not change your firewall |
| 13:33.23 | [TK]D-Fender | So what are you actually meaning by "open up ports"? |
| 13:33.30 | cesurasean | how do i check to see what ports it opened? netstat? |
| 13:33.36 | cesurasean | listening on a port |
| 13:33.43 | [TK]D-Fender | netstat -anp |
| 13:33.48 | WIMPy | yes |
| 13:34.45 | cesurasean | netstat does not show anything |
| 13:34.47 | cesurasean | with asterisk in it |
| 13:35.06 | [TK]D-Fender | What tells you * is actually running? |
| 13:35.21 | [TK]D-Fender | Did you confirm modules are actually loaded? |
| 13:35.36 | [TK]D-Fender | Is this a previously functional install? |
| 13:36.17 | WIMPy | Hmm. Asterisk doesn't have a name here, either. |
| 13:36.55 | cesurasean | i don't know how to confirm the modules are loaded. |
| 13:37.37 | cesurasean | nor is * easy to utilize. |
| 13:38.10 | [TK]D-Fender | netstat -anp|grep asterisk |
| 13:38.16 | [TK]D-Fender | [09:35][TK]D-FenderWhat tells you * is actually running? |
| 13:38.17 | WIMPy | 'modules show' at the *CLI |
| 13:38.31 | cesurasean | oh ok |
| 13:38.34 | cesurasean | forgot about grep |
| 13:38.35 | [TK]D-Fender | module show |
| 13:38.39 | [TK]D-Fender | "no "s" |
| 13:38.39 | cesurasean | sure, says listening |
| 13:38.42 | cesurasean | 0.0.0.0 |
| 13:38.59 | [TK]D-Fender | listening for what? |
| 13:39.19 | WIMPy | wonders why asterisk doesn't have a name here. |
| 13:39.31 | cesurasean | let me paste bin |
| 13:40.32 | cesurasean | http://pastebin.com/ny6vhYEN |
| 13:40.57 | [TK]D-Fender | Asterisk 11.13.1~dfsg-2+b1, Copyright (C) 1999 - 2013 Digium, Inc. and others. |
| 13:40.58 | cesurasean | i suspect it's iptables, which is installed, i just dont know how to admin it. doesn't respond to usual commands in the newer debian versions. |
| 13:41.00 | WIMPy | OH. Interesting. I'm just not allowed to see the process name. |
| 13:41.07 | [TK]D-Fender | Funny it clearly tells you the version right at CLI |
| 13:41.15 | cesurasean | sorry |
| 13:41.26 | [TK]D-Fender | And we see modules there... |
| 13:41.36 | WIMPy | And it also tells you it's from 2013. |
| 13:41.40 | cesurasean | so? |
| 13:41.46 | cesurasean | it's the most stable debian offers. |
| 13:41.50 | [TK]D-Fender | And tells you it is listening on all sorts of ports |
| 13:41.51 | tzafrir | Tip: system.conf lines should not be over 255 characters long |
| 13:41.53 | cesurasean | apparently newer digium is not stable. |
| 13:42.17 | WIMPy | LOL |
| 13:42.26 | [TK]D-Fender | So it looks fine so far |
| 13:42.31 | [TK]D-Fender | Got another question? |
| 13:42.44 | cesurasean | so it's not asterisk messing up? grr |
| 13:42.49 | *** join/#asterisk sparetire_ (~sparetire@unaffiliated/sparetire) |
| 13:42.49 | [TK]D-Fender | messing up? |
| 13:42.51 | [TK]D-Fender | It's there. |
| 13:42.53 | cesurasean | guess i need to hop on over to #iptables |
| 13:42.57 | [TK]D-Fender | We haven't seen a single failure yet |
| 13:43.16 | [TK]D-Fender | Or you could just show us... |
| 13:43.28 | [TK]D-Fender | We tend to have iptables on our servers too... |
| 13:43.37 | [TK]D-Fender | (most). |
| 13:43.41 | cesurasean | so what do i need to add for a default asterisk install? |
| 13:43.47 | cesurasean | does asterisk not come with iptables script to configure it? |
| 13:43.50 | cesurasean | if not, why not? |
| 13:43.53 | [TK]D-Fender | Depends what you are doing before |
| 13:43.58 | [TK]D-Fender | are you blocking things already? |
| 13:44.01 | cesurasean | no |
| 13:44.05 | cesurasean | just a fresh install |
| 13:44.06 | [TK]D-Fender | We have no idea what we're adding to. |
| 13:44.14 | [TK]D-Fender | We don't know what's in your firewall now |
| 13:44.18 | [TK]D-Fender | or if it's even a factor |
| 13:44.20 | cesurasean | its just a vanilla install of debian jessie, bro |
| 13:44.26 | [TK]D-Fender | show us the firewall |
| 13:44.40 | [TK]D-Fender | saying "default" doesn't prove what state it's in. |
| 13:45.09 | [TK]D-Fender | You need to acutally get the packets for the protocols yuo are intending to let * process |
| 13:45.14 | cesurasean | firewall is open |
| 13:45.21 | cesurasean | according to iptables -L, and debian docs |
| 13:45.34 | [TK]D-Fender | docs mean nothing, |
| 13:45.37 | [TK]D-Fender | iptables --list |
| 13:45.40 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 13:45.45 | cesurasean | nothing is in iptables -L |
| 13:45.48 | cesurasean | it's empty |
| 13:46.01 | [TK]D-Fender | And the default policy? |
| 13:46.03 | WIMPy | Where do you try to reach it from? How many other firewalls are there? |
| 13:46.16 | [TK]D-Fender | Show us the dump |
| 13:46.38 | cesurasean | good question. using togglebox.com for cloud hosting. |
| 13:46.47 | cesurasean | they tried to connect and told me issue was at my server level. |
| 13:46.54 | *** join/#asterisk mcargile (~mikec@office1.vicidial.com) |
| 13:46.56 | cesurasean | i was using a 3rd party to ping the 5060 port. |
| 13:47.04 | [TK]D-Fender | how? |
| 13:47.20 | mcargile | how long does it take Asterisk to timeout an AMI connection if it doesnt do anything to keep the connection alive? |
| 13:47.34 | WIMPy | And how did you tell it didn't work? |
| 13:47.55 | WIMPy | mcargile: Why would it time out? |
| 13:47.59 | cesurasean | how do i know which port asterisk listening on? |
| 13:48.08 | [TK]D-Fender | http://pastebin.com/ny6vhYEN |
| 13:48.10 | [TK]D-Fender | PORTS |
| 13:48.12 | [TK]D-Fender | many |
| 13:48.16 | [TK]D-Fender | and we just went through this |
| 13:48.20 | mcargile | Every code example I have seen includes sending a ping to keep the connection alive. |
| 13:48.29 | mcargile | I am just trying to figure out how often I need to do that. |
| 13:48.45 | cesurasean | ok so 5060 is correct |
| 13:48.46 | cesurasean | hmmm |
| 13:49.00 | WIMPy | mcargile: I have never done that. There is no timeout. |
| 13:49.15 | [TK]D-Fender | cesurasean: again what is this "ping" you're talking about exactly? |
| 13:49.26 | cesurasean | i just tried to ping that port |
| 13:49.31 | cesurasean | using ping a port google search |
| 13:49.38 | cesurasean | and found a 3rd party website to ping with |
| 13:49.47 | cesurasean | should i be using telnet or something to connect instead? |
| 13:50.04 | [TK]D-Fender | No |
| 13:50.06 | [TK]D-Fender | BOTH are wrong |
| 13:50.10 | WIMPy | Neither. It's UDP! |
| 13:50.10 | [TK]D-Fender | that is ***TCP*** |
| 13:50.15 | cesurasean | oh |
| 13:50.17 | cesurasean | i see |
| 13:50.22 | [TK]D-Fender | And * is listening on ***UDP*** |
| 13:50.29 | mcargile | WIMPy: are you just listening to the AMI or are you sending actions? |
| 13:50.32 | cesurasean | how come i can't connect 5060 UDP then? |
| 13:50.39 | [TK]D-Fender | Connect how? |
| 13:50.45 | cesurasean | softphone |
| 13:50.47 | [TK]D-Fender | we don't see packets arriving at your server |
| 13:50.49 | cesurasean | zoiper |
| 13:50.55 | [TK]D-Fender | You have not actually shown us your firewall |
| 13:51.02 | cesurasean | firewall is empty |
| 13:51.10 | WIMPy | mcargile: Mostly listening. There will often be days without sending. |
| 13:51.34 | cesurasean | http://pastebin.com/SiXMTGZH |
| 13:52.01 | [TK]D-Fender | "sip set debug on" |
| 13:52.09 | [TK]D-Fender | from *CLI |
| 13:52.24 | [TK]D-Fender | If you see nothing... then nothing is going on. |
| 13:53.20 | cesurasean | it shows some things |
| 13:53.26 | cesurasean | keeps saying wrong password |
| 13:53.29 | cesurasean | with different usernames |
| 13:53.30 | cesurasean | and ports |
| 13:53.47 | *** join/#asterisk martin__1 (~martin@185.32.9.250) |
| 13:54.00 | [TK]D-Fender | from WHO? |
| 13:54.09 | [TK]D-Fender | the IP you're expecting them to come from? |
| 13:54.09 | *** join/#asterisk mjordan (mjordan@nat/digium/x-pbkkyycoezuphzuy) |
| 13:54.09 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:54.16 | cesurasean | no, from the server's ip itself |
| 13:54.20 | martin__1 | we installed asterisk with apt-get, now after chaning database client it seems that the cdr_mysql.so is lost. |
| 13:54.24 | martin__1 | How can we replace it? |
| 13:54.40 | [TK]D-Fender | Time to start showing us... |
| 13:54.44 | WIMPy | mcargile: Where did you find these pings? |
| 13:54.58 | cesurasean | it keeps re-transmitting data |
| 13:55.03 | [TK]D-Fender | Show us |
| 13:55.18 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-pxzkwehvdvdebjbe) |
| 13:55.18 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 13:55.44 | cesurasean | http://pastebin.com/1XWX3NjV |
| 13:56.23 | mcargile | WIMPy: http://search.cpan.org/~greenbean/Asterisk-AMI-v0.2.8/lib/Asterisk/AMI.pm Search for the word ping. |
| 13:56.43 | [TK]D-Fender | Looks like toll-fraud attacks |
| 13:56.44 | mcargile | There is a keep alive in that perl module that you set so it will send one out at an interval |
| 13:56.52 | [TK]D-Fender | And you are accepting unauthed calls which isn't great |
| 13:57.11 | cesurasean | my fresh server is already being attacked? |
| 13:57.48 | [TK]D-Fender | Do you see taht # and IP? |
| 13:57.57 | [TK]D-Fender | They are sending you calls |
| 13:58.10 | WIMPy | mcargile: Yes, it's obviousely about that module. It's probably there so you can connect from remote via a possibly forgetting firewall. But there is definitely no need related to Asterisk itself. |
| 13:58.16 | [TK]D-Fender | If you don't knwo them.... that that is exactly aws it appears : toll-fraud attack |
| 13:58.39 | [TK]D-Fender | <--- SIP read from UDP:37.157.241.124:5070 ---> |
| 13:58.40 | cesurasean | so i need to stop toll fraud attack, and my system will magically start accepting connections? |
| 13:58.40 | [TK]D-Fender | INVITE sip:06660048422885410@108.161.135.66 SIP/2.0 |
| 13:58.59 | [TK]D-Fender | Looking for 06660048422885410 in public (domain 108.161.135.66)it is accepting connections... from THEM |
| 13:59.06 | mcargile | WIMPy: thanks |
| 13:59.09 | *** part/#asterisk mcargile (~mikec@office1.vicidial.com) |
| 13:59.10 | [TK]D-Fender | If your other device isn't getting through then it's your client-end |
| 13:59.27 | [TK]D-Fender | unless the debug you passed was not at a popint in time that client was actually even trying |
| 13:59.32 | cesurasean | does comcast block asterisk ports? |
| 13:59.39 | [TK]D-Fender | So firewall them out to reduce the noise |
| 13:59.55 | [TK]D-Fender | What protocl are you using with Zoiper? |
| 14:00.08 | cesurasean | iax |
| 14:00.18 | [TK]D-Fender | then we aren't looking at the right thing for that |
| 14:00.23 | [TK]D-Fender | that's IAX debug |
| 14:00.24 | [TK]D-Fender | not SIP |
| 14:00.36 | [TK]D-Fender | but block those other people off and fix your config |
| 14:00.46 | [TK]D-Fender | You should not be allowing unauthed calls |
| 14:01.24 | [TK]D-Fender | "iax2 set debug on" <- to see your zoiper attempts |
| 14:01.48 | cesurasean | ok let me digest all of this help first, then i will attempt to fix |
| 14:02.51 | cesurasean | so its possible my iax isnt working but sip is? |
| 14:03.08 | [TK]D-Fender | Anything is "possible", but you are showing us the attempt yet |
| 14:03.18 | [TK]D-Fender | So we have nothing to judge by |
| 14:03.24 | [TK]D-Fender | ran what I told you to and look |
| 14:04.11 | lvlinux | thinks that somebody needs to do some iptables research... and make a blacklist. (in addition to blocking unauth connections to *) |
| 14:04.49 | [TK]D-Fender | lvlinux: FIRST step is fixing *'s acceptance of unauthed calls in the first. |
| 14:05.10 | [TK]D-Fender | Because you'll be chasing after random attackers while each gets their swings in.... |
| 14:05.11 | cesurasean | how do i turn off unauth calls? |
| 14:05.11 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 14:05.33 | [TK]D-Fender | cesurasean: "allowguest=no" under [general] in sip.conf |
| 14:06.06 | martin__1 | I need help kicking about our mysql cdr. If I run module load cdr_mysql.so I recieve that it cannot be found. |
| 14:06.09 | martin__1 | Running asterisk 11.7.0 |
| 14:06.38 | cesurasean | why do they toll fraud anyways? how does that benefit the attackers? porn hackers? |
| 14:06.43 | WIMPy | martin__1: Well, looks like it's not installed then. |
| 14:06.53 | martin__1 | WIMPy: how do I install it? |
| 14:07.02 | WIMPy | cesurasean: Free international calls. |
| 14:07.05 | martin__1 | it's not compiled from source |
| 14:07.18 | cesurasean | oh i see |
| 14:07.42 | lvlinux | cesurasean: in your sip.conf should be allowguest=no |
| 14:07.43 | [TK]D-Fender | cesurasean: TOLL FRAUD. That IS the benifit. |
| 14:07.43 | WIMPy | martin__1: If you didn't make it. I can't help. You have to check with whoever does those packages. |
| 14:08.06 | martin__1 | damn |
| 14:08.15 | cesurasean | ok, now im getting no route to destination |
| 14:08.17 | cesurasean | what's that about? |
| 14:08.22 | [TK]D-Fender | SHOW US |
| 14:08.28 | martin__1 | WIMPy: will it work by taking the cdr_mysql.so from another installation? |
| 14:08.36 | cesurasean | zoiper says it, not sure if that message is on server side |
| 14:08.41 | WIMPy | martin__1: Maybe they removed support for it in favour of ODBC? |
| 14:08.48 | [TK]D-Fender | cesurasean: I gave you a command to run.... |
| 14:08.53 | WIMPy | martin__1: No. |
| 14:09.12 | WIMPy | martin__1: You won't even be able to load that. |
| 14:09.12 | lvlinux | cesurasean: if some dude in nowhereistan can get your * server to make 100 calls to bozoistan at $3.49/min, and they can charge all their "clients" $0.20/min, they are very happy. That's why you get the attacks. |
| 14:10.06 | cesurasean | oh, i see. |
| 14:10.27 | martin__1 | WIMPy: ok, I'll look into odbc |
| 14:10.42 | cesurasean | cool |
| 14:10.49 | cesurasean | so just set debug and make calls to determine issues :P |
| 14:10.51 | cesurasean | easy enough |
| 14:10.54 | lvlinux | martin__1: yes odbc has kindof taken the place of cdr_mysql. Does same thing but works better. |
| 14:11.14 | martin__1 | no configuration has to be made to cdr.conf? |
| 14:11.33 | cesurasean | the softphone isnt kicking back the route error yet though. is that normal? |
| 14:11.45 | lvlinux | martin__1: Yes the config is different. |
| 14:11.51 | [TK]D-Fender | Don't care about the softphone yet |
| 14:12.00 | [TK]D-Fender | we want to see you actually looking on your server |
| 14:12.08 | lvlinux | martin__1: the book tells all about how to set it up. |
| 14:12.17 | cesurasean | can i dial from command line? |
| 14:12.33 | cesurasean | i was able to call myself using the softphone and sip |
| 14:12.42 | cesurasean | but when i make an outbound out the server, it doesnt work |
| 14:12.47 | cesurasean | no destination |
| 14:12.59 | [TK]D-Fender | SHOW US |
| 14:13.01 | cesurasean | 401 unauthorized |
| 14:13.01 | *** join/#asterisk kharwell (kharwell@nat/digium/x-pmyrffgstqyqypgt) |
| 14:13.15 | cesurasean | ok hold on |
| 14:13.23 | [TK]D-Fender | 401 is an auth reject. Has nothng to do with what you're dialing |
| 14:13.25 | [TK]D-Fender | and that is SIP |
| 14:13.33 | [TK]D-Fender | and we were supposed to be looking for IAX |
| 14:14.01 | cesurasean | i changed from iax to sip |
| 14:14.11 | cesurasean | ill get sip working first, then get iax working, i suppose. |
| 14:14.31 | cesurasean | http://pastebin.com/aFL560qp |
| 14:15.06 | cesurasean | digium should implement a pastebin tool for asterisk to post logs here |
| 14:15.19 | cesurasean | would make sharing knowledge easier among the community |
| 14:15.47 | [TK]D-Fender | pastebin.com <- works just fine |
| 14:15.51 | cesurasean | have it email using sendmail |
| 14:15.58 | cesurasean | yeah but would cool |
| 14:16.01 | cesurasean | be* |
| 14:16.03 | [TK]D-Fender | MAIL for debug? |
| 14:16.08 | [TK]D-Fender | COPY/PASTE from SSH |
| 14:16.14 | cesurasean | yeah, like email you an output if you chose to |
| 14:16.21 | cesurasean | why not? |
| 14:16.27 | [TK]D-Fender | it's your system, do it however you want |
| 14:16.39 | [TK]D-Fender | * is a telephony toolkit. What you make of it is up to you |
| 14:17.15 | [TK]D-Fender | <--- SIP read from UDP:68.35.152.116:41135 ---> |
| 14:17.16 | [TK]D-Fender | INVITE sip:2562266599@simpletechnology2.servehttp.com;transport=UDP SIP/2.0 |
| 14:17.17 | cesurasean | going to call everyone and ask for $1 |
| 14:17.20 | [TK]D-Fender | Looking for 2562266599 in DLPN_DialPlan1 (domain simpletechnology2.servehttp.com) |
| 14:17.24 | [TK]D-Fender | SIP/2.0 404 Not Found |
| 14:17.35 | [TK]D-Fender | Your call came in and you have no match in your dialplan to process taht call. |
| 14:18.06 | [TK]D-Fender | And from the looks of it are using Asterisk-GUI which is not supported at all and was abandoned 4 years ago |
| 14:18.23 | cesurasean | which gui is supported now? |
| 14:18.37 | cesurasean | and how can i remove this old bs? |
| 14:18.44 | [TK]D-Fender | FreePBX is still supported. By their own channel of course, not here |
| 14:19.03 | cesurasean | ok so i need to get them to help me remove old gui, and install new one, check. |
| 14:19.10 | *** join/#asterisk Jinxed- (~b0ot@192.160.117.131) |
| 14:19.12 | Jinxed- | Any simple examples of CME(CUCME) trunked with asterisk? Getting a 401 unauthorized from cisco debug |
| 14:19.59 | cesurasean | why does someone use asterisk over any other pbx anyways? |
| 14:20.07 | cesurasean | there are so many different phone systems out there. |
| 14:20.16 | lvlinux | cesurasean: no, you need to remove the gui and keep it removed. |
| 14:20.30 | cesurasean | lol |
| 14:20.39 | WIMPy | cesurasean: Because it's free. |
| 14:20.42 | lvlinux | cesurasean: because asterisk does what we want it to do, fits our requirements, etc. |
| 14:20.55 | WIMPy | As free as Linux: Only if your time is worthless. |
| 14:20.56 | *** join/#asterisk pzn (~pzn@pdpc/supporter/active/pzn) |
| 14:21.12 | lvlinux | lol |
| 14:21.16 | [TK]D-Fender | With * you get actual control |
| 14:21.26 | [TK]D-Fender | * is whatever you want it to be |
| 14:21.29 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-lhcoojjnemgsroha) |
| 14:21.29 | lvlinux | ^^^^^^^^ yes ^^^^^^ |
| 14:21.35 | [TK]D-Fender | for me it's a juke-box and coffee-maker |
| 14:21.48 | lvlinux | coffee maker? |
| 14:21.55 | [TK]D-Fender | Did I stutter? |
| 14:22.01 | cesurasean | i live down the street from digium |
| 14:22.23 | cesurasean | but, i knew about it before they built that building. |
| 14:22.30 | lvlinux | cesurasean: that's cool but not a good reason to use asterisk lol |
| 14:22.48 | cesurasean | well, |
| 14:22.53 | cesurasean | i know the army uses it |
| 14:23.00 | WIMPy | LOL |
| 14:23.02 | pzn | Hi! I'm debugging a firmware for a device that talks sip with asterisk. when the device tries to finish a call with BYE, asterisk answers with code "481 call leg/transaction does not exist". where can I read about what should match in the pkg to finish a call? |
| 14:23.15 | cesurasean | i don't really know much about it though. |
| 14:23.26 | lvlinux | cesurasean: read the asterisk book |
| 14:23.31 | [TK]D-Fender | They wrote a wonderful BOOK |
| 14:23.32 | cesurasean | it's expensive. |
| 14:23.33 | [TK]D-Fender | ~book |
| 14:23.40 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 14:23.40 | [TK]D-Fender | it's FREE ONLINE |
| 14:23.40 | pzn | or how can I "turn on" asterisk debugging for call leg matching? |
| 14:23.40 | lvlinux | cesurasean: it's FREE |
| 14:23.42 | [TK]D-Fender | ^^^ |
| 14:23.46 | mjordan | pzn: the device isn't sending the request with the correct combination of Call-ID, from tag, and to tag |
| 14:23.51 | cesurasean | i want a paperback. |
| 14:23.58 | lvlinux | pzn: "sip set debug on" |
| 14:24.02 | mjordan | hence, we can't match the BYE request to the dialog that was created by the INVITE request |
| 14:24.19 | lvlinux | cesurasean: then go buy one, but in the mean time, read it online |
| 14:24.23 | [TK]D-Fender | $55 isn't that much. |
| 14:24.29 | [TK]D-Fender | if you insist on paperback |
| 14:24.40 | [TK]D-Fender | Trees DIED for this |
| 14:24.42 | cesurasean | i can probably print it myself cheaper. how many pages is it? |
| 14:24.44 | lvlinux | $55! is that how much it costs? goodness that ain't hay... |
| 14:25.11 | lvlinux | but in fairness if it wasn't available free online $55 is well worth it! |
| 14:25.16 | cesurasean | redlobster gave me an old printer i need use for. |
| 14:25.44 | lvlinux | then have at it |
| 14:26.03 | cesurasean | yeah, well that book matches my collection. just on the expensive side. |
| 14:26.13 | lvlinux | you can get it in a single html page, so just have to print one "page" |
| 14:26.32 | [TK]D-Fender | loads another roll into the plotter |
| 14:27.05 | lvlinux | what do you do TK? |
| 14:27.12 | lvlinux | I mean occupation. |
| 14:27.31 | [TK]D-Fender | I am the IT dept for my company |
| 14:27.37 | cesurasean | i fixed a plotter for ITT Tech once. |
| 14:27.46 | lvlinux | ah. Telecom co? |
| 14:27.49 | [TK]D-Fender | So "PC Load Letter Guy" |
| 14:28.08 | lvlinux | lol |
| 14:28.13 | [TK]D-Fender | No, plumbing |
| 14:28.21 | lvlinux | really? |
| 14:28.29 | [TK]D-Fender | Distribution more specifically |
| 14:28.39 | [TK]D-Fender | yup |
| 14:28.53 | lvlinux | haha that makes more sense. I was going to ask---do plumbers have "IT"??? lol |
| 14:29.54 | cesurasean | sure they do. |
| 14:30.13 | lvlinux | you mean when they have to fish a smartphone out of the toilet? |
| 14:30.23 | lvlinux | i guess that qualifies |
| 14:30.36 | WIMPy | LOL |
| 14:30.36 | cesurasean | no, their websites, etc.... phone systems. |
| 14:31.16 | pzn | mjordan, tks! got the point, I captured the packages with wireshark (compared a working with a non-working softphone) |
| 14:32.49 | pzn | mjordan, they both seem similar. "call-id + from" are exactly the same. "to" has a difference in the bye package, it has a ";tag=randon" at the end (in the working device and in the one that is not working). do you have any ideas how can I debug this? |
| 14:34.08 | mjordan | there really isn't much to debug. A 481 means we can't match the request to a transaction; it's a problem with the device, not Asterisk. |
| 14:34.22 | mjordan | I'd push it back to whoever manufactured the device. |
| 14:34.47 | pzn | mjordan, yes... I'm developing the device :-) trying to "turn on" some debug in asterisk that will help me to solve |
| 14:35.19 | mjordan | if you want to see more of what Asterisk is doing, you can turn either 'sip set debug on' or 'pjsip set logger on', crank the core debug up to 5, and watch what happens |
| 14:35.28 | *** join/#asterisk wonderworld (~ww@ip-84-119-184-180.unity-media.net) |
| 14:35.41 | mjordan | But generally this is a relatively low level problem. You should probably go read what the Call-ID, from tag, and to tag actually are in RFC 3261 |
| 14:36.03 | mjordan | Trying to reverse engineer a SIP stack from a working device is not a good way to write a SIP stack :-) |
| 14:36.48 | WIMPy | Do as Steve Jobs told you: Steal one. |
| 14:36.55 | bulkorok | :-D |
| 14:37.13 | mjordan | there are some really good free ones out there. |
| 14:37.20 | bulkorok | sofia |
| 14:37.24 | mjordan | pjsua |
| 14:37.45 | file | sofia is no longer developed, those who use it maintain it themselves |
| 14:38.03 | bulkorok | I forgot about that |
| 14:38.06 | mjordan | yeah, I wouldn't recommend it unless you have the expectation that you are forking and maintaining it |
| 14:38.18 | file | baresip is another up and coming one |
| 14:38.27 | mjordan | resiprocate |
| 14:38.39 | mjordan | hint: don't write your own SIP stack. |
| 14:39.31 | mjordan | unless you really just want to write a SIP stack, in which case, that can be fun. But if the goal is to make a 'phone' or some other actual end user facing product, it's a bad way to go |
| 14:39.47 | bulkorok | http://www.cs.columbia.edu/sip/implementations.html |
| 14:40.42 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 14:41.03 | bulkorok | maybe kamailio is also an option... |
| 14:41.20 | pzn | thanks! I'll take a look about using a ready-to-use sip stack. |
| 14:47.14 | WIMPy | Oh, a TE410P for 11.50 in 6 hours. |
| 14:51.35 | Jinxed- | What causes 401 unauthorized |
| 14:56.39 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 14:58.16 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 15:04.15 | lvlinux | Jinxed-: username, password, wrong domain |
| 15:04.29 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 15:04.30 | lvlinux | wrong peer |
| 15:05.12 | Jinxed- | how would one source number work, and changing the source number causes a 401 |
| 15:06.00 | [TK]D-Fender | Using the From: as the username... |
| 15:06.05 | WIMPy | Because you use it for auth? |
| 15:06.19 | WIMPy | ... which you probably don't want to do. |
| 15:09.24 | Jinxed- | Have two CME trunks dialing the same number, one worked, one didn't, change the source number of the one that didn't to make it like the one that did and now it works too |
| 15:09.53 | Jinxed- | what variable name refers to the source of the call as opposed to number dialed {exten} |
| 15:10.57 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 15:13.41 | Jinxed- | Hmm what is the sipfriends database? |
| 15:14.33 | martin__1 | jeez, this odbc is driving me crazy |
| 15:15.12 | *** join/#asterisk vader- (~Adium@50.232.174.194) |
| 15:17.23 | *** join/#asterisk MissionCritical (~MissionCr@unaffiliated/missioncritical) |
| 15:20.02 | *** join/#asterisk rl1 (~Darwin@80.76.224.9) |
| 15:24.44 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 15:41.47 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 15:43.29 | Jinxed- | What happens after you finish a label line... does it just go to the next line? |
| 15:43.52 | [TK]D-Fender | label is just a text name associated with a priority |
| 15:44.14 | Jinxed- | so after this line: exten => 101,n(am_nauto),Set(Ipaddr=${REALTIME(sipusers,name,${EXTEN})}) |
| 15:44.23 | Jinxed- | if the next line was |
| 15:44.24 | Jinxed- | exten => 101,n,Set(Ipaddr=${CUT(Ipaddr,\,,15)}) |
| 15:44.27 | Jinxed- | it would just go to it |
| 15:44.48 | [TK]D-Fender | yup |
| 15:44.51 | Jinxed- | ok |
| 15:44.52 | Jinxed- | thanks |
| 15:47.02 | Jinxed- | 3 lines to paste - pastebin or chat? |
| 15:48.26 | [TK]D-Fender | pastebin it |
| 15:48.35 | [TK]D-Fender | would have taken less time than asking |
| 15:48.37 | [TK]D-Fender | :p |
| 15:48.44 | Jinxed- | !pastebin |
| 15:48.49 | Jinxed- | ~pastebin |
| 15:48.53 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 15:48.53 | [TK]D-Fender | pastebin.com |
| 15:49.56 | Jinxed- | [TK]D-Fender, doesn't this logic seem backwards |
| 15:49.59 | Jinxed- | http://paste2.org/Vs0P3nnp |
| 15:50.30 | Jinxed- | I thought it was IF(Check)?True:False |
| 15:50.46 | Jinxed- | It seems like they want to ensure Answer-Mode is Auto, but the labels are backwards |
| 15:51.23 | Jinxed- | If it is auto, it adds the header as auto, if it isn't auto then it doesn't do anything |
| 15:52.04 | [TK]D-Fender | Which makes sense |
| 15:52.55 | Jinxed- | oh |
| 15:53.02 | Jinxed- | hmm |
| 15:53.23 | martin__1 | Ok, so I just built asterisk 11 from source. Still no cdr_mysql.so |
| 15:53.25 | martin__1 | Where is it? |
| 15:53.32 | Jinxed- | could you explain a bit more why? |
| 15:54.14 | martin__1 | Well, I want the cdr_mysql.so module. I thought it was default in the asterisk. |
| 15:54.54 | Jinxed- | [TK]D-Fender, If they already checked that the SIP_HEADER Answer-mode was auto, why would they need to SIPAddHeader Answer-Mode auto, wouldn't they only need to do that if the SIP_HEADER Answer-Mode was not Auto? |
| 15:55.07 | [TK]D-Fender | Direct MySQL is a thing of the past. Everything is suposed to be ODBC now |
| 15:55.17 | martin__1 | Well, ODBC works as crap with mariadb |
| 15:55.28 | martin__1 | 5 seems to work, not 10 |
| 15:55.29 | [TK]D-Fender | Jinxed-: Because that should be leading to a call OUT. |
| 15:55.42 | martin__1 | I get crash dumps other funny stuff when using odbc |
| 15:55.42 | [TK]D-Fender | The fact teh call came IN with the header does not make it get included on the call OUT |
| 15:55.45 | [TK]D-Fender | Hence the ADD |
| 15:56.19 | Jinxed- | oh... and if it doesn't have it on the IN it doesn't add it on the OUT |
| 15:57.47 | [TK]D-Fender | Correct |
| 15:58.01 | [TK]D-Fender | this is a poor-man's approach to trying to treat * more like a proxy |
| 15:58.10 | [TK]D-Fender | pass-the-buck |
| 15:59.13 | [TK]D-Fender | martin__1: What version are you running exactly? |
| 16:00.29 | Jinxed- | [TK]D-Fender, is REALTIME a defined asterisk standard? |
| 16:01.13 | Jinxed- | function |
| 16:01.45 | martin__1 | [TK]D-Fender: this is a mariadb issue with odbc |
| 16:01.46 | Jinxed- | Set(Foo=$REALTIME(sipusers,name${EXTEN})) |
| 16:01.52 | martin__1 | it is not asterisk related :/ |
| 16:02.01 | Jinxed- | Set(Foo=$REALTIME(sipusers,name,${EXTEN})) |
| 16:02.23 | [TK]D-Fender | oh well... |
| 16:02.42 | [TK]D-Fender | Jinxed-: yes |
| 16:03.09 | [TK]D-Fender | Jinxed-: And it would be very nice if you would actually call it like a function... |
| 16:03.38 | hirogen1 | anyoen seen it in avay on a phone when you dial an outside line and you click on connect it adds a 1 before so +190208800800 |
| 16:06.24 | *** join/#asterisk igcewieling (~ewieling@ip98-170-211-145.pn.at.cox.net) |
| 16:06.54 | [TK]D-Fender | martin__1: That's up to you and your phone's dialplan |
| 16:07.59 | hirogen1 | thing is her settings are the same as mine |
| 16:08.04 | hirogen1 | so im gonan try an admin tool and reset it |
| 16:23.39 | jeffspeff | anyone have experience connecting asterisk to an NEC system via SIP? |
| 16:37.12 | Kobaz | how do i get callids to display in an attached console |
| 16:39.04 | igcewieling | Kobaz: in the dialplan try using the SIP_HEADER function, something like Set(BOB=${SIP_HEADER(Call-ID)}) |
| 16:39.28 | igcewieling | "sip show channels" might also show what you are looking form. |
| 16:39.29 | Kobaz | not in the dialplan |
| 16:39.31 | Kobaz | in all messages |
| 16:39.45 | Kobaz | ie: like using -T to turn on console timestamping |
| 16:39.51 | igcewieling | Kobaz: then the answer, as far as I can tell you cannot. |
| 16:40.02 | igcewieling | you could enable SIP debug I guess. |
| 16:40.20 | [TK]D-Fender | "All messages" |
| 16:40.24 | [TK]D-Fender | nope |
| 16:40.25 | Kobaz | yeah i'm looking at the logger source and it looks like display_callids option is only for controlling the log file output |
| 16:40.36 | Kobaz | trying to find where it does the console output |
| 16:41.29 | igcewieling | I take the CALL-ID and log it to the CDR, but that's not what you are looking for. |
| 16:41.57 | Kobaz | nope |
| 16:47.25 | igcewieling | good luck with that. |
| 16:49.46 | *** join/#asterisk DragonAzul (~DragonAzu@187.208.12.193) |
| 16:52.37 | *** join/#asterisk DynamicFail (~b0ot@212.144.253.35) |
| 16:53.40 | Kobaz | got it |
| 16:53.43 | Kobaz | that wasn't too bad |
| 16:53.50 | *** join/#asterisk DynamicFail (~b0ot@212.144.253.35) |
| 16:54.13 | Kobaz | i'll put up a patch for it |
| 16:55.45 | Kobaz | <PROTECTED> |
| 16:55.47 | Kobaz | pretty cool |
| 16:56.20 | Kobaz | [TK]D-Fender: now i can! |
| 16:56.39 | [TK]D-Fender | now YOU can and WE still can't ;) |
| 16:56.45 | Kobaz | aw |
| 16:56.49 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 16:56.50 | Kobaz | poor we |
| 17:00.09 | Kobaz | so, i hope you're all excited |
| 17:00.20 | Kobaz | i'm gonna work on my massive logging update patch this time around |
| 17:00.42 | Kobaz | more informational and more consistent logging across the board |
| 17:10.11 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 17:20.43 | igcewieling | That is not the SIP Call-ID |
| 17:20.59 | igcewieling | That is the Asterisk channel name. *totally* different. |
| 17:21.14 | *** join/#asterisk jcarlos (~quassel@89.129.149.44) |
| 17:22.08 | jcarlos | One question, please. How can I change the "asterisk" word in the From: field of the INVITE method? |
| 17:22.29 | jcarlos | Talking about sip channel, of course |
| 17:22.43 | igcewieling | Isn't the channel name is already shown in the CLI? like: -- Executing [s@sub-record-check:11] ExecIf("SIP/139-0000333b", "0?Return()") in new stack |
| 17:25.05 | [TK]D-Fender | you shouldn't get that unless you are missing callerid entirely |
| 17:25.15 | [TK]D-Fender | jcarlos: ^ |
| 17:25.31 | Kobaz | igcewieling: yeap. i'm not talking about sip call id |
| 17:25.51 | Kobaz | and it's not the channel name either |
| 17:25.57 | Kobaz | it's the call id |
| 17:26.33 | orn | does asterisk need a restart after changing TOS settings? |
| 17:27.03 | orn | never mind |
| 17:27.03 | Kobaz | igcewieling: core call id... ie: [C-00000003] |
| 17:27.40 | orn | needs to run as root or have libpcap installed, apparently |
| 17:27.56 | Kobaz | igcewieling: and heh, i'm so used to my own logging that your log line looked really weird |
| 17:32.21 | jcarlos | [TK]D-Fender: Yes, that was the case. I thought the 404 response I get was because of the presence of asterisk word. I have changed it and no difference :( http://paste.debian.net/318760/ |
| 17:33.25 | [TK]D-Fender | User-Agent: DSL Router/DSL Router-00.96.315 |
| 17:33.34 | jcarlos | I'm trying to connect Asterisk to the SIP server in my router to do outbound calls |
| 17:33.37 | [TK]D-Fender | INVITE sip:9xxxxxxxx@192.168.0.1 SIP/2.0 |
| 17:33.43 | [TK]D-Fender | They don't like the # you are dialing |
| 17:34.26 | jcarlos | I don't now why, because I can use the router with my android phone and no problem with the number |
| 17:34.43 | [TK]D-Fender | From: "Dockterisk" <sip:*108@192.168.0.2>;tag=as20aabca0 |
| 17:34.48 | [TK]D-Fender | or it's 404'ing your ID |
| 17:35.15 | [TK]D-Fender | Giving I'm use the packets from your phone have the username in the from |
| 17:35.20 | [TK]D-Fender | so try fixing your peer |
| 17:35.25 | jcarlos | It is the same From: field that I have when dialing from a SIP client in my Android phone |
| 17:35.29 | [TK]D-Fender | fromuser=THEUSERNAME |
| 17:35.30 | jcarlos | And it works in that case |
| 17:35.55 | [TK]D-Fender | *108 I'm suspecting is not the username |
| 17:36.00 | jcarlos | Yes, it is |
| 17:36.10 | [TK]D-Fender | rather unusual |
| 17:36.38 | jcarlos | Take into account that those usernames are for the SIP server in the router |
| 17:36.59 | jcarlos | They are not "global" usernames |
| 17:37.24 | jcarlos | Everybody with the same router (and the same SIP server) wil have to use the same usernames |
| 17:37.39 | jcarlos | They are of the form *101, *102, *103, ... |
| 17:37.48 | igcewieling | Kobaz: Call-ID: 4e63309a2ab3c1c91d6da5bc72a58a65@10.0.0.254:5060 |
| 17:37.53 | igcewieling | That is a CallID. |
| 17:39.16 | [TK]D-Fender | well it's either the number or the user |
| 17:39.22 | [TK]D-Fender | those are the things that get 404'd |
| 17:39.59 | jcarlos | [TK]D-Fender: The thing I don't understand is that when I call using my Android SIP client (using directly the SIP server in the router) I get a 401 and the negotiation proceed correctly |
| 17:40.20 | jcarlos | But if it is my asterisk box the one to contact with the SIP server in the router, I get a 404 |
| 17:40.39 | igcewieling | is the sip server the same as the asterisk box? |
| 17:40.50 | [TK]D-Fender | I'd need to see 2 real full comms for comparison |
| 17:40.55 | *** join/#asterisk quintana (~quintana@modemcable094.94-70-69.static.videotron.ca) |
| 17:41.10 | jcarlos | [TK]D-Fender: I will send. One moment |
| 17:41.16 | Kobaz | igcewieling: that's a sip call id |
| 17:41.25 | Kobaz | igcewieling: that's not a core asterisk call id |
| 17:41.30 | *** join/#asterisk italorossi (~Adium@187.60.66.11) |
| 17:41.38 | Kobaz | igcewieling: very different |
| 17:43.52 | igcewieling | *nod* I must be really out of touch. Never heard anyone call those "call IDs" |
| 17:45.57 | Kobaz | logger.conf [general] display_callids |
| 17:46.18 | Kobaz | i think it's only in 11 and up, it may be in 10, don't know offhand |
| 17:53.54 | igcewieling | Kobaz: must be Asterisk 12 or 13 or undocumented in Asterisk 11. That string does not exist in the sample configs of Asterisk 11 |
| 17:54.04 | Kobaz | it's not in the sample configs |
| 17:54.07 | Kobaz | the default is enabled though |
| 17:54.17 | Kobaz | it's in 11 |
| 17:54.18 | Kobaz | i'm using it now |
| 17:55.55 | igcewieling | someone might consider documenting it. |
| 17:56.15 | igcewieling | I've wished we could see the logging ID in the CLI for years. |
| 17:56.36 | Kobaz | well i just added it in my own branch |
| 17:56.42 | Kobaz | it doesn't log to the cli by default |
| 17:57.47 | igcewieling | Ah, sorry, I thought you said it was included with Asterisk 11+ |
| 17:58.13 | *** part/#asterisk dummys (~dummys@unaffiliated/dummys) |
| 17:58.41 | Kobaz | call ids are included, yes |
| 17:58.48 | Kobaz | the cli logging, i just added that now |
| 17:58.53 | igcewieling | Kobaz: but not documented |
| 17:58.59 | Kobaz | it's not documented but |
| 17:59.01 | Kobaz | you can still use it |
| 17:59.05 | Kobaz | look at your /var/log/asterisk/full |
| 17:59.17 | Kobaz | and it's enabled by default |
| 17:59.18 | igcewieling | I am referring *ONLY* to the CLI. |
| 17:59.23 | Kobaz | okay, yeah |
| 17:59.27 | igcewieling | I am NOT referring to log files. |
| 17:59.29 | Kobaz | that's not available by default in any version |
| 18:00.54 | igcewieling | I wonder why they use the exact same term as SIP uses. I'd imagine it could lead to significant amounts of confusion. |
| 18:01.04 | Kobaz | i wouldn't think |
| 18:01.08 | Kobaz | it's called a sip call id |
| 18:01.16 | Kobaz | and even if you said call id, it's still specific to sip |
| 18:01.29 | Kobaz | if you're using iax, or dahdi, or webrtc, a sip call id doesn't apply anyway |
| 18:06.35 | *** join/#asterisk u0m3 (~u0m3@89.120.204.99) |
| 18:17.48 | *** join/#asterisk areski (~areski@80.174.128.118.dyn.user.ono.com) |
| 18:19.10 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 18:49.28 | cusco | hi folks |
| 18:50.23 | cusco | we have a sip account from service provider, and now outbounds start to get 'Forbidden'. We did not change a thing. Support tells us that we're sending auth in the first INVITE, and that we should send it on the second frame |
| 18:52.25 | cusco | current peer details in sip.conf: http://paste.debian.net/318764/ |
| 18:54.50 | [TK]D-Fender | nat = force_rport,comedia |
| 18:55.11 | [TK]D-Fender | no sane provider is ever behind NAT. This should be "no" |
| 18:55.20 | cusco | right... |
| 18:55.25 | cusco | missed that |
| 18:55.35 | cusco | bu that is not the issue, is it? |
| 18:55.44 | [TK]D-Fender | Should also probably set the USER in there |
| 18:55.52 | [TK]D-Fender | fromuser=USERNAME, and pass ICD in RPID |
| 18:55.58 | [TK]D-Fender | sedrpid=yes |
| 18:56.05 | [TK]D-Fender | sendrpid=yes |
| 18:56.32 | [TK]D-Fender | Which you seem to have at least |
| 18:56.38 | [TK]D-Fender | now use FROMUSER with it |
| 18:56.56 | [TK]D-Fender | And then if that fails, show us the actual attempt |
| 18:56.59 | cusco | ok.. |
| 18:59.22 | cusco | [TK]D-Fender: still failled: http://paste.debian.net/318765/ |
| 18:59.40 | [TK]D-Fender | Contact: <sip:309984718@10.100.100.8:5060> |
| 18:59.45 | [TK]D-Fender | Yeah... clearly bad NAT setup |
| 18:59.49 | [TK]D-Fender | failed the basics... |
| 18:59.59 | [TK]D-Fender | You are passing them your PRIVATE IP |
| 19:00.46 | cusco | yea thats why I had nat on, I guess .. can't remember |
| 19:00.47 | [TK]D-Fender | On a side note: Asterisk PBX 1.8.12.0 |
| 19:00.53 | cusco | yes... I know :( |
| 19:01.06 | [TK]D-Fender | You are more that 20 releases behind ... on a DEAD BRANCH |
| 19:01.25 | [TK]D-Fender | Add that to the list after fixing your SIP configs |
| 19:01.43 | cusco | it has been on my list for like over a year |
| 19:01.58 | cusco | also, upgrading debian wheezy to jessie |
| 19:08.37 | *** join/#asterisk areski (~areski@80.174.128.38.dyn.user.ono.com) |
| 19:23.43 | rrittgarn1 | is cdr_adaptive_odbc supported in 13? doing some tests upgrading from 11 to 13 and having issues with custom columns not populating values from dialplan |
| 19:24.27 | rrittgarn1 | reloaded the module and i see the custom columns, but when setting their values via SET(CDR(customcolumn)=value) i don't get that value stored in cdr records |
| 19:25.35 | *** join/#asterisk MaliutaLap (nikolai@unaffiliated/maliuta) |
| 19:26.43 | drmessano | Is the module loaded? |
| 19:30.59 | rrittgarn1 | cdr_adaptive_odbc.so Adaptive ODBC CDR backend 0 Running core |
| 19:31.01 | rrittgarn1 | yessir |
| 19:32.21 | drmessano | I would say it's supported then |
| 19:32.54 | rrittgarn1 | touchez |
| 19:50.42 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 19:59.40 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-172.cust.bezeqint.net) |
| 20:01.43 | rrittgarn1 | I went and set a static value to store into one of my custom columns, and that works fine, but the dialplan doesn't. Syntax wise: same => n,Set(CDR(userid)=1); should work right? |
| 20:02.23 | robmal | IDK |
| 20:02.31 | robmal | There's a wiki. |
| 20:02.36 | robmal | They have examples there. |
| 20:02.48 | robmal | It's like a manual. |
| 20:03.13 | rrittgarn1 | https://wiki.asterisk.org/wiki/display/AST/CDR+Storage+Backends?focusedCommentId=31097306#comment-31097306 <--- Like this one that says its missing? |
| 20:04.30 | robmal | Ooooh. |
| 20:04.52 | robmal | Meh, ok, looks like i'll help you ;-) |
| 20:05.40 | robmal | So, you've got adaptive working, its detecting a new column? |
| 20:05.45 | rrittgarn1 | correct |
| 20:05.57 | rrittgarn1 | and i can statically set values in the config file, and those make it into my DB |
| 20:06.02 | rrittgarn1 | all the standard columns make it too |
| 20:06.25 | rrittgarn1 | but when i try to set values in dialplan (specifically in the H extension - as worked in 11) - the values are not making it |
| 20:06.40 | rrittgarn1 | looking at the debug log the column isn't referenced either |
| 20:06.42 | robmal | Oh noes, h exten. |
| 20:07.05 | robmal | Ok. Do a NoOp just before Set to verify if the value is still there. |
| 20:09.44 | rrittgarn1 | strange. I set the value and two lines later the value isn't there... |
| 20:10.01 | robmal | *magic* |
| 20:11.21 | rrittgarn1 | k... so i can't set cdr values in the H exten anymore? |
| 20:12.21 | robmal | AFAIR the values set during the call should still be there, but imho using extension h to do anything is asking for trouble. |
| 20:13.03 | igcewieling | H was never a valid extension, AFIK. However the h extension is valid. |
| 20:13.07 | rrittgarn1 | I was not aware that it was that problematic |
| 20:13.22 | rrittgarn1 | and yes, referring to the 'h' exten, not H |
| 20:13.32 | robmal | What are you trying to achieve? |
| 20:13.43 | igcewieling | rrittgarn1: Have you considered using hangup handlers instead?? |
| 20:13.43 | *** join/#asterisk Inf0r (uid2810@gateway/web/irccloud.com/x-qwmggxvpntesukbv) |
| 20:14.39 | rrittgarn1 | after the call is processed, I update the cdr with a unique user id for the person on my system that made the call, said ID is looked up externally via a SQL query based on the channels that were connected. |
| 20:15.13 | robmal | Why after? You know who made the call in the beginning. |
| 20:16.07 | rrittgarn1 | hadn't addressed that honestly. it was easier at the time to just do cleanup at the end when originally implemented. |
| 20:16.10 | rrittgarn1 | @igcewieling, kind of thought that's what i was doing - please enlighten me into better ways |
| 20:24.16 | robmal | shines light on setting cdr values before hangup. |
| 20:24.59 | igcewieling | rrittgarn1: https://wiki.asterisk.org/wiki/display/AST/Hangup+Handlers+Specification |
| 20:25.25 | rrittgarn1 | thank you kind sirs |
| 20:27.40 | tm1000 | in conferences. Is there a way for a menu option to jump to another menu. So you could have more than 10 options |
| 20:29.39 | robmal | dialplan_exec? |
| 20:29.55 | robmal | It could playback something and wait for digits. |
| 20:30.43 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 20:30.54 | tm1000 | robmal: super gross |
| 20:31.05 | tm1000 | was wondering if there was something simplier. heh :-) |
| 20:31.05 | robmal | Ok, so the answer is no. |
| 20:31.35 | robmal | No, seriously, did you try setting a menu item with two digits? |
| 20:31.47 | tm1000 | robmal: no. was going to try it next |
| 20:32.20 | robmal | I think it'll work. |
| 20:35.08 | *** join/#asterisk theron (~theron@2620:10d:c090:200::c:694a) |
| 20:41.34 | tm1000 | robmal: doesnt work. thats ok though. thanks for your help |
| 20:42.42 | robmal | Good to know. |
| 20:43.22 | *** join/#asterisk superscrat (asanders@nat/digium/x-hxajdeyekblakijk) |
| 20:43.45 | *** join/#asterisk Voyage (~user1@unaffiliated/voyage) |
| 20:44.15 | Voyage | How to name the recorded file as <number-called-date.mp3>? |
| 20:44.24 | Voyage | using MixMonitor() |
| 20:44.32 | robmal | Voyage! Welcome back! |
| 20:44.39 | Voyage | robmal, I missed you too |
| 20:44.45 | robmal | I see you still can't read manuals! :-) |
| 20:45.19 | Voyage | I am searching for one |
| 20:45.27 | Voyage | that tells me that |
| 20:45.36 | robmal | Ok. |
| 20:45.42 | robmal | Pick a letter. |
| 20:45.51 | Voyage | 'a' |
| 20:45.56 | MaliutaLap | robmal: 5 |
| 20:45.59 | Kobaz | igcewieling: yay hangup handlers :) |
| 20:46.21 | robmal | Voyage: ________________A__ |
| 20:46.34 | Voyage | b |
| 20:46.37 | robmal | *bzzt* |
| 20:46.43 | Voyage | c |
| 20:46.45 | robmal | *bzzt* |
| 20:46.47 | Voyage | d |
| 20:46.51 | robmal | *bzzt* |
| 20:46.52 | Voyage | e |
| 20:46.54 | robmal | Penality: 5min |
| 20:46.59 | Voyage | lol |
| 20:48.20 | *** join/#asterisk malcolmd (malcolmd@pdpc/sponsor/digium/malcolmd) |
| 20:48.34 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 20:48.42 | Voyage | DNID: Dialed Number IDentifier? |
| 20:50.29 | Voyage | what is 'dialed number' named in variable and dialplan scripting? |
| 20:50.35 | Voyage | the outgoing dialed number |
| 20:51.15 | Voyage | ${CALLERID(dnid)} * - Dialed Number Identifier ? |
| 20:51.59 | robmal | K, penality over, current game: ________________A__ |
| 20:52.30 | Voyage | e |
| 20:52.36 | robmal | ________________A_E |
| 20:52.42 | Voyage | l |
| 20:52.58 | robmal | _____________L_A_E |
| 20:53.02 | robmal | Sorry. |
| 20:53.08 | igcewieling | ${EXTEN} almost always is the DNID. Possible excepions which come to mind are if you are using a PRI |
| 20:53.10 | robmal | ____________L_A_E |
| 20:53.14 | robmal | Damn. |
| 20:53.38 | robmal | ____________L__A_E |
| 20:53.40 | robmal | Now. |
| 20:54.08 | Voyage | igcewieling, ${EXTEN} * - Current extension |
| 20:54.30 | Voyage | igcewieling, doesnt makes sense how somehow get the number that I called from it |
| 20:55.46 | igcewieling | The number you are called from is CallerID, which is not the DNID. |
| 20:55.59 | Voyage | right. I dont want it either |
| 20:56.01 | Kobaz | source number (callerid) and destination number (exten/dnid) |
| 20:56.05 | Voyage | I want the callee number |
| 20:56.25 | Kobaz | callee = destination |
| 20:56.57 | Voyage | Kobaz, ok, so (exten/dnid) can be fetched by either ${EXTEN} or ${CALLERID(dnid)} * - Dialed Number Identifier ? |
| 20:57.14 | Kobaz | yeap |
| 20:57.24 | igcewieling | My recommendation is to set a dialplan variable to the value of ${EXTEN} right where the call would enter the dialplan. That way if ${EXTEN} changes, for example using a gosub or macro, you'll still have the original digits. |
| 20:57.50 | Voyage | igcewieling, ok. |
| 20:57.53 | igcewieling | Voyage: usually, unless you are on PRI |
| 20:58.09 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 20:58.11 | Voyage | Iam using AMi |
| 20:58.20 | Voyage | to originate |
| 20:58.21 | robmal | We know. |
| 20:58.36 | Voyage | robmal, pick a letter |
| 20:58.43 | robmal | F |
| 20:58.43 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 20:58.46 | igcewieling | I track EXTEN, DIALSTATUS and HANGUPCAUSE that way in my dialplan |
| 20:58.48 | Voyage | F______________ |
| 20:58.51 | robmal | U |
| 20:58.56 | Voyage | FU____________ |
| 20:58.57 | robmal | C |
| 20:59.05 | Voyage | ok, I dont want to be banned |
| 20:59.08 | robmal | ;-) |
| 20:59.11 | robmal | Poor game. |
| 20:59.22 | Voyage | at least you know what you deserve |
| 20:59.29 | Voyage | just kidding |
| 21:01.12 | robmal | </3 |
| 21:04.26 | *** join/#asterisk Oatmeal (~Suzeanne@75-103-145-152.ccrtc.com) |
| 21:08.48 | Voyage | igcewieling, Kobaz exten => _1NXXXXXXXXX,1,MixMonitor(${EXTEN}.wav) worked but exten => _1NXXXXXXXXX,1,MixMonitor(${CALLERID(dnid)}.wav) did not |
| 21:10.02 | robmal | Oh noes. |
| 21:12.48 | *** join/#asterisk mac_ified (~mac_ified@67-9-150-210.res.bhn.net) |
| 21:16.40 | *** join/#asterisk EOIP (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 21:17.19 | igcewieling | Have you tried ${CALLERID(DNID)} ? |
| 21:19.19 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:20.04 | [TK]D-Fender | You have no callerid |
| 21:20.09 | [TK]D-Fender | Of course it will fail |
| 21:20.32 | *** join/#asterisk robink_ (~quassel@unaffilated/robink) |
| 21:21.07 | robmal | thinks [TK]D-Fender is a Jedi. |
| 21:21.40 | [TK]D-Fender | No, if I could force-choke people at great distances this shannel would be a lot slimmer... |
| 21:21.50 | robmal | :-) |
| 21:21.55 | robmal | I wish you could. |
| 21:22.28 | MaliutaLap | [TK]D-Fender++ |
| 21:22.55 | MaliutaLap | force-choking - they ultimate LART |
| 21:32.58 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 21:38.16 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 21:44.35 | *** join/#asterisk theron (~theron@2620:10d:c090:200::d:fea3) |
| 22:02.49 | Voyage | igcewieling, yes, I tried ${CALLERID(DNID)} and it did not worked |
| 22:03.01 | Voyage | because I think I dont have a caller id in dialplan |
| 22:03.03 | [TK]D-Fender | <[TK]D-Fender> You have no callerid |
| 22:03.03 | [TK]D-Fender | <[TK]D-Fender> Of course it will fail |
| 22:03.06 | Voyage | I used ami origination |
| 22:03.21 | Voyage | right |
| 22:04.28 | Voyage | what sthe best format to record voice for asterisk? |
| 22:04.38 | [TK]D-Fender | the same format as your call is in |
| 22:04.39 | Voyage | wav is too big |
| 22:04.42 | robmal | Klingon |
| 22:05.43 | Voyage | [TK]D-Fender, I think default is GSM in * but the helloworld was choosen by * as .slin |
| 22:06.09 | [TK]D-Fender | There is no such thing as "default". |
| 22:06.16 | [TK]D-Fender | and "think" shouldn't be coming into this |
| 22:06.31 | [TK]D-Fender | You should have set this in your peer yourself and it shouldn't be a guess |
| 22:06.42 | Voyage | hm |
| 22:06.50 | Voyage | in my peer? |
| 22:07.04 | [TK]D-Fender | <[TK]D-Fender> You should have set this in your peer yourself and it shouldn't be a guess |
| 22:07.11 | Voyage | in my peer? |
| 22:07.15 | [TK]D-Fender | YES |
| 22:07.20 | [TK]D-Fender | Did I stutter? |
| 22:07.26 | Voyage | super |
| 22:07.29 | robmal | N-no. |
| 22:07.31 | [TK]D-Fender | Is there an echo in here? |
| 22:07.43 | robmal | Is there an echo in here? |
| 22:07.46 | [TK]D-Fender | could have sworn he installed an echo canceller |
| 22:08.05 | Voyage | robmal, _____T |
| 22:09.04 | robmal | Voyage: IDIOT has 5 letters, but nice to meet you. I'm Robert. |
| 22:09.40 | Voyage | You are TREBOR or Trouble |
| 22:10.12 | Voyage | [TK]D-Fender, ______H |
| 22:10.24 | robmal | I'm doing my best to make you leave this channel and do some research on your own. |
| 22:10.37 | Voyage | robq not gonna happen unless I get banned |
| 22:10.48 | Voyage | [TK]D-Fender, pick a letter |
| 22:11.30 | *** join/#asterisk talntid (~talntid@173-160-189-58-Washington.hfc.comcastbusiness.net) |
| 22:12.11 | talntid | anyone here know if i can chanspy all the members of a queue, but only on the calls that are within that queue? so if they are in 2 queues... Queue1 and Queue2... and I want to spy on all Queue1 calls? |
| 22:12.22 | [TK]D-Fender | And the reason you think I'm going to play some ridiculous game is? |
| 22:12.51 | [TK]D-Fender | talntid, There is no spying on "queue calls", there is only "spying on channels" |
| 22:13.16 | [TK]D-Fender | talntid, You're going to have to script up whatever kind of filtering you want in choosing which channels are actually queueu calls |
| 22:13.29 | talntid | gotcha. hmmmmmm |
| 22:13.44 | robmal | http://hackrr.com/2013/freepbx/how-to-monitor-certain-queues-with-chanspy/ |
| 22:13.47 | robmal | This. |
| 22:14.57 | [TK]D-Fender | hrm |
| 22:15.01 | [TK]D-Fender | loks like I missed a var |
| 22:15.20 | Voyage | playback () does not gives control to next line while background() does. background is useful for user, for e.g, to press 1,2,3 etc and still keep the background sound going. The problem in this that if user presses something, the next line is run (which is hangup() ). I want to make sure the background sound was fully played before hangup? Any solution else than wait()? |
| 22:15.33 | Voyage | wait(seconds) |
| 22:15.35 | robmal | Holocaust |
| 22:15.53 | [TK]D-Fender | grp - Only spy on channels in which one or more of the groups |
| 22:15.53 | [TK]D-Fender | <PROTECTED> |
| 22:15.53 | [TK]D-Fender | <PROTECTED> |
| 22:15.55 | [TK]D-Fender | yup |
| 22:16.22 | [TK]D-Fender | So if you set the group before you jump into the queue and it inherits right (and the rest of the stars align), yup, that could do it |
| 22:16.39 | robmal | <- had the right thing in bookmarks |
| 22:17.07 | [TK]D-Fender | Voyage, If you want it played in full then use Playback. |
| 22:17.28 | [TK]D-Fender | The entire point of Backgound IS to be interrupted |
| 22:17.37 | Voyage | [TK]D-Fender, but I also want to get number options |
| 22:18.04 | [TK]D-Fender | then get them AFTER the playback |
| 22:18.11 | Voyage | hm |
| 22:18.27 | rrittgarn1 | or playback a certain amount of your audio before switching to background |
| 22:18.35 | rrittgarn1 | slice the audio at a pause between words |
| 22:18.40 | robmal | But you could use background and set some exten => 1,1,Hangup() |
| 22:18.50 | [TK]D-Fender | No, he said "fully played" |
| 22:18.55 | [TK]D-Fender | that means NO interruption |
| 22:19.10 | robmal | So playback it is. |
| 22:19.14 | [TK]D-Fender | Get your input afterwards |
| 22:19.48 | Voyage | ok |
| 22:25.14 | Voyage | How can I write certain things to log file or some customFile.txt. I am not talking about verbose(0 or NoOp() which are for console printf |
| 22:26.04 | robmal | logger.conf |
| 22:26.29 | [TK]D-Fender | Something that makes a call outside of * |
| 22:26.42 | [TK]D-Fender | "core show function SHELL" |
| 22:26.44 | Voyage | is this the only way left? https://wiki.asterisk.org/wiki/display/AST/Logging |
| 22:26.50 | [TK]D-Fender | "core show application system" |
| 22:26.55 | Voyage | ok |
| 22:26.58 | [TK]D-Fender | "no |
| 22:27.03 | Voyage | am. hmm |
| 22:27.15 | [TK]D-Fender | "core show application agi" <- which you are NOT ready for |
| 22:27.41 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 22:27.44 | drmessano | ~book |
| 22:27.44 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 22:27.49 | MaliutaLap | I can't remember this much hand holding in a long time |
| 22:27.58 | drmessano | DAYS of it |
| 22:28.11 | drmessano | Like TK Built the whole thing |
| 22:28.31 | drmessano | For a contract job the guy is getting paid to do |
| 22:28.38 | MaliutaLap | drmessano: I think we need to send [TK]D-Fender on that force-choking course |
| 22:28.44 | robmal | I'm pretty sure [TK]D-Fender is dying and he's trying to fix all his bad deeds in one deal. |
| 22:29.20 | MaliutaLap | robmal: when this kind of things happen we all die a little inside |
| 22:29.33 | drmessano | No doubt Vonage over there is being paid for this custom thing. His demands are too specific for it not to be a job |
| 22:29.35 | robmal | Not all of us. I just eat more popcorn. |
| 22:29.42 | drmessano | So he's had someone else do it for him |
| 22:29.49 | drmessano | Yay for work ethic |
| 22:29.50 | [TK]D-Fender | drmessano, "so... do you think management will be looking to hir a replacement for the position immediately?" |
| 22:30.11 | [TK]D-Fender | </classic> |
| 22:30.20 | drmessano | He'll be back on IRC the first time he needs to change it |
| 22:30.42 | robmal | He'll be paid to use IRC = great success |
| 22:30.58 | drmessano | There's a line between HELP and whatever the hell this is |
| 22:31.06 | MaliutaLap | I hate people who can't read docs |
| 22:31.18 | drmessano | Or do the actual work? |
| 22:31.29 | drmessano | And it's "won't" |
| 22:31.33 | drmessano | Not "can't" |
| 22:31.40 | MaliutaLap | probably both |
| 22:31.45 | drmessano | He just wants to finish the job and get paid |
| 22:32.03 | drmessano | Expending as little effort as possible |
| 22:32.29 | robmal | For free! |
| 22:32.36 | drmessano | And NOW |
| 22:32.50 | robmal | Which is awesome. |
| 22:33.26 | MaliutaLap | because not understanding how something you've done works is a good thing? |
| 22:33.49 | Voyage | [TK]D-Fender, no "core show function SHELL" "core show application system" "core show application agi" is not something I want. I want to write some things to a file.txt on certain events. eg. call connected to number <number here> (this might be possible by AGI status) but, what about "user pressed number <number here>" |
| 22:33.52 | robmal | It's like going to a brothel and crying in the entrance until you get laid :-) |
| 22:33.54 | drmessano | Him and I had it out in #freepbx ... I told him #freepbx was support for a SPECIFIC PROJECT. He told me "I am here for COMMUNITY". Still don't know WTF that means |
| 22:34.26 | drmessano | Like Ivan Drago in Rocky IV |
| 22:34.36 | drmessano | I DO FOR ME |
| 22:34.47 | robmal | drmessano: You should support his lack of intelligence. Community means caring for the weaker. |
| 22:35.08 | [TK]D-Fender | Voyage, And what I gave you was the solution to that |
| 22:35.15 | drmessano | Oh, so he's retarded? Why didn't someone say so |
| 22:36.08 | robmal | I'm pretty sure he is, no other option [TK]D-Fender would be so patient. |
| 22:36.11 | drmessano | Arguing on IRC is like winning a gold medal in the Special Olympics. If you win, you're still retarded |
| 22:36.26 | [TK]D-Fender | tries to force-choke anyway..... |
| 22:36.26 | Voyage | [TK]D-Fender, If I am doing calls via AGI, what are my options to know call status and number that user pressed? |
| 22:36.42 | drmessano | Oh AGI now? |
| 22:36.46 | [TK]D-Fender | You know what they pressed because you read it in as input |
| 22:36.58 | [TK]D-Fender | And as I said, you are ready for AGI |
| 22:36.59 | drmessano | Just use Asterisk -x a LOT |
| 22:37.02 | [TK]D-Fender | NOT* |
| 22:37.15 | *** join/#asterisk robink_ (~quassel@unaffilated/robink) |
| 22:37.17 | Voyage | [TK]D-Fender, yes, I am getting the input stream fine. that is ONE option. do I have any other? |
| 22:37.30 | [TK]D-Fender | Do you need another? |
| 22:37.34 | robmal | [TK]D-Fender: Don't say that. You let him skip dialplan logic straight to AMI, i'm sure you'll be able to help him with AGI. |
| 22:37.36 | drmessano | Yep, sniff some glue |
| 22:37.38 | MaliutaLap | something; something; kill -9 -1 |
| 22:37.44 | Voyage | I am just learning things and expanding my mind. [TK]D-Fender |
| 22:37.51 | drmessano | MaliutaLap: lol |
| 22:37.59 | drmessano | Expanding what?! |
| 22:38.16 | MaliutaLap | drmessano: well there is no other way for it to go ;) |
| 22:38.20 | Voyage | [TK]D-Fender, you said I am not ready for AMI [TK]D-Fender> "core show application agi" <- which you are NOT ready for |
| 22:38.32 | robmal | :-] Fun++ |
| 22:38.37 | drmessano | Hahahhahah |
| 22:38.42 | [TK]D-Fender | double checks his echo canceller.... |
| 22:38.45 | drmessano | AGI != AMI |
| 22:38.57 | robmal | Voyage: Yes, if you're not ready for one level up you should try going to level 3 :-) |
| 22:38.58 | Voyage | ah type |
| 22:39.02 | Voyage | typo |
| 22:39.12 | drmessano | SPELLING IS A BIZZATCH |
| 22:39.15 | [TK]D-Fender | NOT A TYPO |
| 22:39.20 | [TK]D-Fender | agi != ami |
| 22:39.31 | drmessano | So is being paid to do a job and getting someone else to do it |
| 22:39.36 | robmal | Voyage: Go ARI! |
| 22:39.59 | drmessano | Just use the QBASIC API |
| 22:40.04 | drmessano | AQI |
| 22:40.11 | drmessano | 10 call |
| 22:40.18 | drmessano | 20 goto 10 |
| 22:40.24 | drmessano | ROBODIALER |
| 22:41.04 | [TK]D-Fender | LANA!!!!! |
| 22:41.05 | [TK]D-Fender | ... |
| 22:41.09 | [TK]D-Fender | DANGER ZONE |
| 22:41.29 | drmessano | lol |
| 22:41.38 | Sircle | Voyage: AMI = Asterisk manager API. Thats what you are using (the stream thing told me). I have been observing you since some time. Try to google as much as you can before asking here |
| 22:42.00 | MaliutaLap | [TK]D-Fender: dude - phrasing |
| 22:42.12 | *** join/#asterisk SpeakerToMeat (~SpeakerTo@prgmr/customer/SpeakerToMeat) |
| 22:42.35 | drmessano | Sircle: he can't find anything on Google |
| 22:42.46 | drmessano | Apparently Google is out of AMI |
| 22:42.50 | MaliutaLap | seriously? are we not doing phrasing anymore? |
| 22:42.50 | drmessano | AND TOMATOES |
| 22:44.04 | Voyage | [TK]D-Fender, I was expecting to log things via dialplan (when the call had reached it). The only thing to get from AMI was status code (whether the call was connnected or rejected/busy etc) |
| 22:44.16 | Voyage | by log I mean write to a file |
| 22:44.26 | Voyage | not verbose() or noop() |
| 22:44.28 | [TK]D-Fender | This has nothing to do with AMI anymore |
| 22:44.34 | [TK]D-Fender | You're in your channel |
| 22:44.34 | Voyage | that was my simple question |
| 22:44.37 | [TK]D-Fender | the call is processing |
| 22:44.39 | Voyage | ok |
| 22:44.44 | [TK]D-Fender | and I gave you the means by which you can do this |
| 22:45.38 | Voyage | [TK]D-Fender, ok, I can do all the above via AMI stream. Even what button was pressed. But is there any other option? |
| 22:45.43 | Voyage | option/way |
| 22:45.50 | [TK]D-Fender | no, you can't |
| 22:45.58 | Voyage | NO other way? |
| 22:45.58 | [TK]D-Fender | and this has nothing to do with AMI |
| 22:46.12 | robmal | Lies. |
| 22:46.13 | [TK]D-Fender | Stop talking about AMI |
| 22:46.23 | Voyage | nothing to do with AMI? I though you said the stream tells status etc |
| 22:46.25 | Voyage | ok, |
| 22:46.29 | Voyage | ignores AMI |
| 22:46.32 | [TK]D-Fender | NO I DIDN'T |
| 22:46.37 | robmal | Voyage: Telnet to localhost 5038 and enter credentials from /etc/asterisk/manager.conf |
| 22:46.42 | [TK]D-Fender | YOU think != I said |
| 22:46.57 | [TK]D-Fender | You are coming up with random "facts" in your head |
| 22:47.11 | [TK]D-Fender | This is not what's documented, nor what we said |
| 22:47.19 | Voyage | <[TK]D-Fender> You know what they pressed because you read it in as input |
| 22:47.48 | [TK]D-Fender | Yes. Now go use a dialplan app that reads in input |
| 22:47.57 | MaliutaLap | why don't we have a tactical missile system for this channel? |
| 22:47.59 | [TK]D-Fender | I sure did NOT say "AMI" in there |
| 22:48.01 | robmal | Voyage: Yes you can! If you have an account admin with secret=admin in manager.conf just telnet to 5038 and watch the events. It's all there. [TK]D-Fender just doesn't want to help you with this. |
| 22:48.55 | robmal | MaliutaLap: I suggested war with #freepbx and it seems Voyage came from there. We could have avoided all of this just by launching a preliminary strike. |
| 22:49.00 | Voyage | robmal, why would i telnet to 5038 when I am already having my app AMI to connect to 5038 and can see the input/output stream |
| 22:49.08 | Voyage | those has events too |
| 22:49.17 | robmal | Voyage: Manager listens on port 5038 |
| 22:49.40 | robmal | Telnet there and you'll see all the events [TK]D-Fender doesn't want you to know about. |
| 22:49.40 | MaliutaLap | robmal: now we're just going to have to go nuclear on it |
| 22:49.43 | Voyage | I was expecting to log things via dialplan (when the call had reached it). The only thing to get from AMI was status code (whether the call was connnected or rejected/busy etc) |
| 22:50.10 | robmal | MaliutaLap: I can't wait. |
| 22:50.47 | robmal | Voyage: Telnet to 5038 |
| 22:50.50 | robmal | Username: xxx |
| 22:50.54 | robmal | Secret: qqq |
| 22:50.56 | robmal | <PROTECTED> |
| 22:51.03 | [TK]D-Fender | robmal, Stop |
| 22:51.16 | robmal | [TK]D-Fender: You started this circus :-) |
| 22:51.30 | [TK]D-Fender | No, I just picked up the whip that was laying over in the corner |
| 22:51.39 | MaliutaLap | kinky |
| 22:51.41 | Voyage | robmal, I dont want to see ALL events. I want to do thing programatically. write x if y happens in dialplan execution or AMi origination success/failure |
| 22:51.43 | robmal | ;-)))) |
| 22:51.46 | Voyage | [TK]D-Fender, and others, I didnt told you to write scripts for me but I am unable to understand a clear answer. Can we be specific? |
| 22:52.31 | MaliutaLap | robmal: fixing up my tax return is looking really attractive right about now |
| 22:52.34 | *** join/#asterisk theron (~theron@173-28-183-190.client.mchsi.com) |
| 22:52.43 | robmal | No, you made [TK]D-Fender answer all your dumb questions. It's amazing, because usually he's rather harsh for newbies. Good job. |
| 22:53.39 | robmal | MaliutaLap: My accountant informed me i have to give back all the tax office gave back as a return because i had to correct an invoice from last year. Love taxes. |
| 22:54.19 | MaliutaLap | robmal: my mother is an accountant - she did my return, and I found a glaring error. Now I have to fix it up |
| 22:54.31 | MaliutaLap | robmal: can't fire her, she's my mother |
| 22:54.31 | robmal | Even worse ;-) |
| 22:54.32 | Voyage | :( |
| 22:55.20 | robmal | Voyage: Don't be sad. This channel didn't have a puppet before so you're a lucky one. |
| 22:56.28 | MaliutaLap | robmal: don't try and get him to use puppet to automate this :P |
| 22:56.59 | robmal | Good idea, let's send him to #puppet |
| 22:58.12 | MaliutaLap | robmal: no, I maintain a presence there too :P |
| 22:58.53 | robmal | So #docker? |
| 22:59.42 | MaliutaLap | yeah, that'll do |
| 22:59.55 | robmal | Voyage: Guys at #docker know shit, go there. |
| 23:01.55 | [TK]D-Fender | heads out to practice.... |
| 23:02.12 | Voyage | [TK]D-Fender, thanks |
| 23:02.32 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 23:15.22 | *** part/#asterisk kharwell (kharwell@nat/digium/x-pmyrffgstqyqypgt) |
| 23:16.17 | *** join/#asterisk wonderworld (~ww@ip-84-119-184-180.unity-media.net) |
| 23:20.28 | Voyage | <PROTECTED> |
| 23:22.22 | robmal | Voyage: Dude. Focus now. Focus hard. /etc/asterisk/logger.conf |
| 23:24.08 | Voyage | robmal, ok, but I planned to to it in real time when the call would be in progress etc |
| 23:24.11 | Voyage | like Set(FILE(/tmp/foo.txt,,,al)=bar) |
| 23:25.17 | robmal | I don't seem to get the point of that. |
| 23:27.32 | Voyage | robmal, nice question. I want to keep a record for each call at one place and nearby lines in the file.txt. e.g I am originating a call via AMI. Now only AMI can know if the call was connected or not. and only dialplan and AMI can know what number was pressed. I just wanted to do things in nearby proximity |
| 23:27.48 | Voyage | or maybe I am not an expert and assumed things wrong? |
| 23:28.45 | robmal | I almost died from laughing from that 'not an expert' part |
| 23:29.05 | robmal | Ok, back to the drawing board. |
| 23:29.10 | robmal | What do you want to achieve? |
| 23:29.32 | Voyage | dont worry, you will die (some day) |
| 23:29.59 | Voyage | robmal, am. log number, connected: in another file, not connected/reject/hangup in an other file |
| 23:30.22 | Voyage | number pressed 1, 2 etc. in the connected.txt file (with the number) |
| 23:30.26 | robmal | Ok, so you're looking for DIALSTATUS |
| 23:30.33 | Voyage | a couple of things more (those are in dialplan) |
| 23:30.58 | Voyage | robmal, ok, i got dIALStatus from AMI status events input stream. cool. what next? |
| 23:31.33 | Voyage | user pressed 1 |
| 23:32.13 | robmal | AMI has DTMFsomething events. |
| 23:32.28 | Voyage | so you want all to be fetched via AM? |
| 23:32.32 | Voyage | AMI* |
| 23:33.02 | robmal | No, i would do it totally different. |
| 23:33.07 | Voyage | ok, nothing bad in it. I agree. It can be done |
| 23:33.14 | Voyage | oh, what would you do: |
| 23:33.18 | Voyage | ? |
| 23:33.31 | Voyage | any why |
| 23:34.11 | robmal | You're making an automated dialout machine, right? With IVR when the other end answers? |
| 23:34.18 | Voyage | correct |
| 23:34.23 | Voyage | IVR? |
| 23:34.42 | robmal | Interactive Voice Rsomething. |
| 23:34.57 | Voyage | kind of |
| 23:35.39 | robmal | So you need to know dialstatus to log when the other end answers, and you'll get a shit load of false positives due to voicemail. |
| 23:36.29 | Voyage | ya, absolutely correct. but there is no way to distinguish voicemail pickup and manual answer by human |
| 23:36.43 | robmal | Of course there is. |
| 23:36.50 | Voyage | really? what is? |
| 23:37.13 | robmal | Lots of things learn you must, young one. |
| 23:37.41 | Voyage | just give me the subject and I will read about it |
| 23:38.01 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 23:38.11 | Voyage | robmal, thats even more interesting. If its a voicemail, I would just hangup |
| 23:38.14 | robmal | I think my stomach just bursted from laughing. |
| 23:38.28 | robmal | How the hell you'll know if it's voicemail? |
| 23:38.34 | Voyage | tell me when you have finished laughing |
| 23:38.41 | robmal | I'm done. |
| 23:38.45 | Voyage | Voyage> ya, absolutely correct. but there is no way to distinguish voicemail pickup and manual answer by human |
| 23:38.47 | Voyage | <robmal> Of course there is. |
| 23:39.15 | robmal | So, yes, i know how to determine if it's VM or a human. |
| 23:39.28 | Voyage | how? |
| 23:39.43 | robmal | I get paid to tell that. |
| 23:39.51 | Voyage | how much you want? |
| 23:40.11 | robmal | I'm not in your price range. |
| 23:40.24 | Voyage | try me (reasonably) |
| 23:40.44 | robmal | $500 just for the idea. |
| 23:41.12 | Voyage | I would prefer to hire you per hour and count how many times i did __ with you |
| 23:42.02 | robmal | No problem, $500/h |
| 23:42.24 | Voyage | and I can do whatever in that hour? |
| 23:42.41 | robmal | No sex, sorry. |
| 23:42.53 | robmal | Not that kind of service. |
| 23:42.54 | Voyage | torture? |
| 23:43.07 | robmal | If it's done by a woman, maybe. |
| 23:43.19 | Voyage | :) |
| 23:43.40 | Voyage | ok anyways. are you going to tell me how to distinguish voicemain answer with manual answer? |
| 23:44.07 | robmal | Sure. |
| 23:44.25 | Voyage | ok, tell me |
| 23:44.30 | robmal | https://www.paypal.me/robmal |
| 23:45.10 | Voyage | ok, what would you have chosen? if not AMI status events? |
| 23:45.48 | robmal | There is no other way to determine if the call was answered or not. |
| 23:46.02 | Voyage | which no other way? |
| 23:46.06 | Voyage | What do you mean. |
| 23:46.11 | *** join/#asterisk [NC] (~nc@rv1.sabius.net) |
| 23:46.29 | Voyage | No, The call is an "answered call" either answered by a human or a voicemail |
| 23:46.40 | robmal | Yes. |
| 23:46.49 | Voyage | so how do you differenciate? |
| 23:46.55 | Voyage | if that number is not owned by you |
| 23:47.43 | robmal | I'm a magician. With ~98% success rate. |
| 23:49.01 | Voyage | if you are saying http://www.voip-info.org/wiki/view/Asterisk+cmd+AMD those are just guesses |
| 23:49.38 | robmal | No, that's not it, not even close. |
| 23:50.00 | Voyage | ok. I dont want to hear it if you dont want to tell it |
| 23:50.14 | Voyage | ok, what would you have chosen? if not AMI status events? |
| 23:50.43 | *** join/#asterisk ruied (~ruied@bl7-216-43.dsl.telepac.pt) |
| 23:50.57 | robmal | I do use dialstatus to determine if the call was answered. There's no other way. But determining if it was answered by a human is *magic* |
| 23:52.02 | talntid | from asterisk cli, can i check a variable for a specific channel? |
| 23:52.18 | robmal | Yes. core show channel xfiojoidjwqoidjwqoi |
| 23:52.36 | Voyage | robmal, where will you get the 'dialstatus'? |
| 23:52.54 | talntid | awesome. i was doing sip show channel |
| 23:52.55 | talntid | thanks |
| 23:53.35 | robmal | Voyage: During a call after Dial() a var called dialstatus is set. |
| 23:53.51 | robmal | But it's exactly the same as your AMI event. |
| 23:54.01 | robmal | ANSWER or NOANSWER, BUSY, that shit. |
| 23:54.15 | Voyage | ok |
| 23:54.36 | talntid | ok, so.... I have this: http://pastebin.com/fUU1Vsbd ... if you check SpyGroup |
| 23:54.36 | craigify | Voyage, did you read the docs on Dial() |
| 23:54.37 | craigify | ? |
| 23:54.43 | robmal | craigify: He didn't. |
| 23:54.44 | talntid | it shows AgentsInc-Main: |
| 23:54.48 | craigify | why not??? |
| 23:54.55 | Voyage | just for learning, how to write that dialstatus to a file.txt? |
| 23:54.57 | robmal | craigify: He's retarded a bit. |
| 23:54.58 | craigify | it's all right there, in lots of details. all the switches |
| 23:55.02 | talntid | ChanSpy(,g(AgentsInc-Main:)) |
| 23:55.11 | craigify | the return variables |
| 23:55.11 | Voyage | like Set(FILE(/tmp/foo.txt,,,al)=bar) ? ? robmal |
| 23:55.13 | talntid | but if I do this, I get more channels than just the spygroup=AgentsInc-Main: |
| 23:55.16 | craigify | or rather the dialplan variables that get set |
| 23:55.28 | craigify | everything you ever want to know is all documented |
| 23:55.32 | craigify | Asterisk has pretty decent documentation |
| 23:55.44 | robmal | craigify: Voyage can't read manuals or documentation. |
| 23:56.07 | craigify | language barrier? |
| 23:56.24 | Voyage | craigify, robmal is just being an asshole. |
| 23:56.39 | robmal | Not just now, all the time. |
| 23:56.50 | Voyage | Ya, its a childhood issue you have |
| 23:57.11 | robmal | Nah, i'm fine with my childhood, i just don't understand people like you :-) |
| 23:57.18 | Voyage | craigify, so I write something to a file from dialplan by this? Set(FILE(/tmp/foo.txt,,,al)=bar) ? ? thats the only thing I found in docs |
| 23:57.41 | robmal | craigify: He's your puppet now, enjoy. |
| 23:57.42 | Voyage | robmal, intelligence or thinking problem you have? |
| 23:58.06 | Voyage | robmal, so, before me, you were his puppet? |
| 23:58.27 | igcewieling | Strange, "core show function FILE" has lots of examples. |
| 23:58.52 | Voyage | igcewieling, yes. Set(FILE(/tmp/foo.txt,,,al)=bar) . I pasted one. Just asked if thats the only way to log things? |
| 23:59.01 | robmal | Voyage: None, but from the tests i've taken i seem to have negative emotional inteligence. So i don't understand people. Especially the dumb ones. Don't be offended by my attitude, sarcarsm is my way of letting off steam. |
| 23:59.29 | craigify | http://stackoverflow.com/questions/4244911/normal-file-i-o-operations-in-asterisk |
| 23:59.43 | igcewieling | Voyage: spend some time looking at the output of "core show functions" and "core show applications", it will save you much time. |
| 23:59.45 | Voyage | robmal, I know, some people are not normal and some people are abnormal. You are later |
| 23:59.59 | robmal | Voyage: I know. |