| 00:00.05 | Voyage | so, can anyone tell me whats wrong here and what -exactly- is wronge and what to replace it with ? http://pastie.org/10509971 |
| 00:00.40 | Voyage | <PROTECTED> |
| 00:00.58 | *** join/#asterisk agoodm (~alan@host81-149-38-69.in-addr.btopenworld.com) |
| 00:02.21 | Voyage | quotes "<talntid> Voyage, arguing with __those of us who do know what you are trying to learn___, won't cause us to be useful to you" |
| 00:02.22 | agoodm | Hi; I've got a weird issue where asterisk 11.20 is using 100% of one core when the following things are true: selinux enforcing and started via systemd using the init script that comes with the asterisk source. If any of these things are not true asterisk behaves normally. Have tried with sample config. |
| 00:06.31 | MaliutaLap | Voyage: the "Channel" should be "SIP/flowrate" - the "Exten" should be "1321xxxxxx7" |
| 00:07.09 | Voyage | [TK]D-Fender> SIP/+1326546547@flowroute <--------------- this IS the "Channel:" from AMI Originate |
| 00:07.23 | Voyage | MaliutaLap, so one of you is mistaken? |
| 00:08.17 | WIMPy | Don't use that @ syntax. Make it sip/flowroute/+1.... |
| 00:08.26 | MaliutaLap | Voyage: depends on what you are trying to achieve |
| 00:09.00 | Voyage | WIMPy, what should be out.println("Exten: then? |
| 00:09.03 | MaliutaLap | are you trying to place a call from an outgoing channel to a local extension? |
| 00:09.10 | WIMPy | And context/exten/priority are the point in your dilplan to hadle that channel. |
| 00:09.12 | Voyage | MaliutaLap, "make a call of course" . its says originate. |
| 00:09.25 | Voyage | MaliutaLap, to a normal phone |
| 00:09.43 | Voyage | WIMPy, what exactly in my case? |
| 00:09.56 | WIMPy | I don't know. |
| 00:10.11 | Voyage | thats what my question is |
| 00:10.20 | WIMPy | Some place that does what you want to do with that call. |
| 00:10.29 | MaliutaLap | Voyage: and you only have the one user in sip.conf? |
| 00:11.28 | MaliutaLap | Voyage: because your internal phone needs to be defined somewhere for this to actually bridge a call |
| 00:13.01 | WIMPy | Is bridging wanted? |
| 00:13.30 | MaliutaLap | WIMPy: the point of originating the call is that both ends are terminated right? |
| 00:13.40 | Voyage | I dont have a soft/hardphone |
| 00:13.49 | WIMPy | Not neccessarily. |
| 00:14.32 | WIMPy | And from what I recall from yesterday the ide was to put them in some IVR. I didn't really read so far today. |
| 00:15.08 | WIMPy | Or actually the day before yesterday and yesterday here. |
| 00:15.40 | MaliutaLap | Voyage: read http://forums.asterisk.org/viewtopic.php?t=75153 |
| 00:16.03 | MaliutaLap | and substitute for the IVR or whatever you need |
| 00:16.07 | *** join/#asterisk jonno11 (~Jon@cpc1-walt12-2-0-cust582.13-2.cable.virginm.net) |
| 00:16.44 | Voyage | to proof that my configs are ok, originate SIP/+1321xxxxxx7@flowroute extension s@outgoing <- worked |
| 00:20.55 | *** join/#asterisk DragonAzul (~DragonAzu@187.208.19.58) |
| 00:23.07 | *** join/#asterisk u0m3 (~u0m3@89.120.204.99) |
| 00:29.07 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-pgcvbgceqpdfugcm) |
| 00:31.11 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 00:39.52 | *** part/#asterisk agoodm (~alan@host81-149-38-69.in-addr.btopenworld.com) |
| 00:54.20 | *** join/#asterisk [NC] (~nc@rv1.sabius.net) |
| 00:59.28 | *** join/#asterisk cyford (~support@c-73-137-1-6.hsd1.ga.comcast.net) |
| 00:59.39 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 01:00.27 | *** join/#asterisk tedstriker (~tedstrike@mig01416328395.demon.co.uk) |
| 01:15.41 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-86-39.hsd1.or.comcast.net) |
| 01:21.40 | *** join/#asterisk cesurasean (~cesurasea@c-68-35-152-116.hsd1.al.comcast.net) |
| 01:21.56 | cesurasean | anyone mind helping me configuring asterisk to use callcentric? |
| 01:22.03 | cesurasean | seems impossible for a newbie. |
| 01:25.38 | Voyage | MaliutaLap, WIMPy robmal talntid [TK]D-Fender the issue was priviliage to AMI user. I had to add 'originate' and I missed that. THIS was not mentioned in * wiki. Or I missed that. and I also had to delete "+" before the number in code. |
| 01:26.04 | Voyage | cesurasean, youwill be asked to read the docs and a book, then re read it. then ask. |
| 01:27.22 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-172.cust.bezeqint.net) |
| 01:28.46 | *** join/#asterisk KNERD (~KNERD@netservisity.com) |
| 01:38.04 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 01:43.04 | *** join/#asterisk monsterco (~monsterco@70.51.53.82) |
| 01:50.59 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-172.cust.bezeqint.net) |
| 01:51.48 | *** join/#asterisk craigify (~Adium@ip184-186-18-123.no.no.cox.net) |
| 01:58.40 | *** join/#asterisk robink (~quassel@unaffilated/robink) |
| 02:00.12 | *** join/#asterisk SamBrereton (~SamBreret@119.63.221.214) |
| 02:18.13 | *** join/#asterisk Jesterboxboy (~Thunderbi@61.170.223.235) |
| 02:21.07 | Jesterboxboy | is there a way to get information about ongoing calls in the asterisk dialplan? specifically i need to check if a call with a certain src number is already underway. |
| 02:21.08 | Jesterboxboy | Is it possible to do this from the dialplan or do i need to use an external script with "core show channels concise" or similar command? |
| 02:23.22 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-86-39.hsd1.or.comcast.net) |
| 02:27.55 | *** join/#asterisk almostworking (~almostwor@unaffiliated/almostworking) |
| 02:30.01 | *** part/#asterisk hjf (~hjf@unaffiliated/hjf) |
| 02:31.23 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 02:36.27 | *** join/#asterisk robink_ (~quassel@unaffilated/robink) |
| 02:43.47 | *** join/#asterisk nny (601eebc0@gateway/web/freenode/ip.96.30.235.192) |
| 02:44.45 | *** join/#asterisk tristero (~al.f.zero@unaffiliated/transfinite) |
| 02:45.04 | nny | anyone have a modern install guide for oslec with Centos6 and asterisk 13? |
| 02:48.40 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 02:54.49 | WIMPy | Isn't it included in Centos? |
| 02:57.09 | nny | Found this http://docs.tzafrir.org.il/dahdi-linux/#_oslec but the damn Rhino driver crashed the server. Waiting on a restart. Not a fan of RHINO hardware :\ |
| 02:57.46 | nny | but yeah it's in the kernel source as well as the git repo listed there |
| 03:06.04 | nny | WIMPy: thanks for answering, more fun later |
| 03:06.49 | *** join/#asterisk snadge (~snadge@unaffiliated/snadge) |
| 03:07.05 | *** join/#asterisk bmg505 (~leon@196-215-47-175.dynamic.isadsl.co.za) |
| 03:07.09 | snadge | how does one show transcoding stats for g729? |
| 03:07.20 | snadge | i know about core show translation.. but thats not really telling me what im after |
| 03:07.41 | snadge | i just want to get an idea of how much cpu is being chewed up translating codecs |
| 03:08.13 | snadge | or even just how many channels are being transcoded |
| 03:10.58 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 03:11.19 | [TK]D-Fender | look at your channels then |
| 03:11.22 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-86-39.hsd1.or.comcast.net) |
| 03:15.28 | snadge | module show gives me a use count |
| 03:15.43 | [TK]D-Fender | that is not "looking at your channels" |
| 03:15.51 | snadge | so if it says 20.. im presuming that means the codec module is being used 20 times |
| 03:15.52 | [TK]D-Fender | "sip show channels" <---------- |
| 03:15.57 | [TK]D-Fender | big print |
| 03:16.05 | snadge | yeah.. that tells me what format the channel is in.. but not necessarily if its being transcoded ? |
| 03:16.18 | snadge | eg. if its just being passed through, or actually being converted |
| 03:16.27 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 03:16.35 | [TK]D-Fender | and "g726 show" give you the list of encoders/decoders in use |
| 03:16.42 | [TK]D-Fender | g729* |
| 03:16.55 | [TK]D-Fender | When you look at the channel it is BRIDGED to you know |
| 03:17.16 | [TK]D-Fender | but "g729 show" already gives you an exact number |
| 03:18.19 | snadge | i get no such command for that |
| 03:18.30 | snadge | im guessing 1.6 is too old |
| 03:20.46 | [TK]D-Fender | g729 <tab> |
| 03:20.51 | [TK]D-Fender | it's been there since forever |
| 03:24.45 | Jesterboxboy | i need to check if a call from the same source is already in progress, is this possible from the dialplan? with "core show channels concise" i found the necessary information, but i am at a loss at how to solve this problem |
| 03:24.56 | [TK]D-Fender | 1.6 is no longer supported. |
| 03:25.02 | [TK]D-Fender | 1.8 is no longer supported |
| 03:25.05 | [TK]D-Fender | upgrade NOW |
| 03:25.16 | snadge | this is going to happen when i get back from thailand :D |
| 03:25.46 | snadge | we have one server in the core network that has been upgraded to asterisk 11 cert |
| 03:26.11 | snadge | but some changes need to be made to the web based billing sytem.. to accommodate the change in the asterisk database format etc |
| 03:26.13 | [TK]D-Fender | Jesterboxboy, "core show application ChanIsAvail", "core show function DEVICE_STATE", "core show functions like GROUP", take your pick |
| 03:26.15 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-86-39.hsd1.or.comcast.net) |
| 03:50.09 | Jesterboxboy | [TK]D-Fender thansk for the input, it seems that GROUP_MATCH_COUNT seems to accomodate my needs the best. |
| 03:50.09 | Jesterboxboy | when i do a show channels concise i get this when a call is ongoing: |
| 03:50.09 | Jesterboxboy | http://codepad.org/Tj2wmFM3 |
| 03:51.00 | Jesterboxboy | [TK]D-Fender iam not quite sure how i would filter out calls that i.e. have the (Outgoing Call) text and some specific other string like the 190403346806001 in it? |
| 03:51.53 | [TK]D-Fender | Clarify EXACTLY what you mean by "same source |
| 03:52.21 | [TK]D-Fender | because that is flagged as the same device from what we see |
| 03:52.38 | [TK]D-Fender | Actually... |
| 03:52.57 | [TK]D-Fender | ALMOST the same |
| 03:52.58 | [TK]D-Fender | SIP/mgw34.ertgo.cc |
| 03:53.04 | [TK]D-Fender | SIP/mgw34 |
| 03:53.08 | [TK]D-Fender | What the 2 peers? |
| 03:53.16 | [TK]D-Fender | At which point I'd GROUPCOUNT them both |
| 03:54.28 | Jesterboxboy | mhm i want to check if a call is underway that has the same string (in this case the 190403346806001) in the channel information. |
| 03:54.34 | Jesterboxboy | its some account id i use, |
| 03:54.37 | Jesterboxboy | the peer is always the same in this case |
| 03:54.56 | [TK]D-Fender | well you can already see it in CLI |
| 03:54.58 | [TK]D-Fender | which means you can get it in a SHELL |
| 03:55.08 | [TK]D-Fender | and you can grep it |
| 03:55.09 | [TK]D-Fender | So that's the solution for that |
| 03:55.44 | [TK]D-Fender | You could also use AMI in an external script. There just isn't a dialplan app for that specific bit without an intermediaty step |
| 03:56.30 | Jesterboxboy | mhm, i am missing something, how would i then tell asterisk in the dialplan to act according to that? |
| 03:56.45 | *** join/#asterisk DragonAzul (~DragonAzu@187.208.54.19) |
| 03:56.49 | Jesterboxboy | i thought i could do something like Gotoif(${GROUP_MATCH_COUNT}>=1)) |
| 03:58.42 | Jesterboxboy | okay so the best way would be to use an external script with AMI, there just grep if the pattern occurs, and return false or true to asterisk |
| 03:59.26 | Jesterboxboy | mhm i just wanted to avoid that as i havent yet looked into AMI :D |
| 03:59.33 | Jesterboxboy | but thanks for the input |
| 04:01.00 | [TK]D-Fender | SHELL + grep works, AMI works |
| 04:01.17 | [TK]D-Fender | How you get that specific number... will check if anything else comes into play |
| 04:01.24 | [TK]D-Fender | but this could be a single SHELL() |
| 04:01.53 | Jesterboxboy | AHH, now the light goes on |
| 04:02.07 | Jesterboxboy | i forgto about the SHELL() Application |
| 04:03.20 | Jesterboxboy | i guess in this case its just a grep combined with a wc -l or something then i have the occurences |
| 04:07.57 | Jesterboxboy | thanks, that was what i was missing |
| 04:38.48 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 05:32.44 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-hqeczwmsjlorlynl) |
| 05:55.37 | *** join/#asterisk K0HAX (michael@shellhost.home.englehorn.com) |
| 06:09.29 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 06:43.31 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 06:47.05 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 06:56.34 | *** join/#asterisk DonkeyDong (~donkey@static.168.209.46.78.clients.your-server.de) |
| 07:44.33 | *** join/#asterisk pchero_work (~pchero@109.70.54.56) |
| 07:45.14 | rl1 | Hmm... When asterisk receives a 302 Moved message, the new call leg it creates is a Local channel in the same context right? Why do all my "transfered" calls have 0 billsec in cdr??? |
| 07:46.46 | rl1 | Hmm.. apparently that is only true for SQL cdr |
| 07:46.51 | rl1 | not for the csv's one |
| 07:50.50 | *** join/#asterisk evil_gordita (robert@ip70-188-63-173.rn.hr.cox.net) |
| 07:58.38 | *** join/#asterisk Zogot (~Adium@185.21.52.255) |
| 08:07.07 | *** join/#asterisk ThomasKeller (~Thomas@vmx.ethz.ch) |
| 08:10.47 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:13.01 | *** join/#asterisk bulkorok (~Adium@89.245.151.228) |
| 08:18.43 | rl1 | is it a known bug? |
| 08:19.05 | rl1 | I use a legacy 1.8.23.0 |
| 08:24.23 | *** join/#asterisk areski (~areski@80.174.128.113.dyn.user.ono.com) |
| 08:24.40 | *** join/#asterisk bluemerlin (~bluemerli@office-nat.westpoint.gradwell.net) |
| 08:25.47 | *** join/#asterisk Ikikay (Ikikay@ADijon-156-1-28-162.w90-39.abo.wanadoo.fr) |
| 08:31.53 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 08:37.17 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 08:38.53 | *** join/#asterisk CeBe (~CeBe@a81-14-224-229.net-htp.de) |
| 08:42.00 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 09:02.55 | *** join/#asterisk wonderworld (~ww@ip-84-119-184-180.unity-media.net) |
| 09:07.52 | wdoekes | rl1: the whole cdr record system is a known bug |
| 09:09.16 | wdoekes | if however the sql cdrs differ from the csv ones, that sounds like something that may be possible to fix (but maybe it already has, you should check the issue tracker) |
| 09:19.58 | rl1 | nope I was wrong. The csv and sql don't differ. Apparently I have to add "/n" into Dial command when transferring a call |
| 09:30.21 | rl1 | alright. How do I do that? I have a phone that sends 301 and asterisk does something like "Dial(Local/DIVERT_NUMBER@incoming_context)" how do I make it add "/n" into that dial command? |
| 09:42.09 | *** join/#asterisk D30 (~D30@222.127.13.226) |
| 09:44.14 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 09:49.45 | rl1 | fucking shit |
| 09:50.03 | rl1 | that's why you shouldn't use asterisk as your main softswitch platform |
| 09:54.13 | rl1 | man for real??? do i really have to deal with this shit. oh fuck :( way to hate my job |
| 09:54.49 | Chainsaw | Is there someone we need to call? Has your medication run out? |
| 09:56.12 | rl1 | yeah call 911 and tell them someone on the internet is having a seizure because of the asterisk CDR bug |
| 10:02.56 | phpboy | anybody here managed to get webrtc to work? |
| 10:19.33 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 10:21.23 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 10:26.06 | *** join/#asterisk areski (~areski@106.Red-83-37-156.dynamicIP.rima-tde.net) |
| 10:28.36 | *** part/#asterisk dummys (~dummys@unaffiliated/dummys) |
| 10:29.45 | Zogot | phpboy: yep |
| 10:30.29 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 10:30.31 | Zogot | question: in agi, if I give another command after STREAM FILE , depending on how the file im streaming is, it wont play the next STREAM FILE's i give it |
| 10:30.57 | Zogot | so STREAM FILE /example/file then STREAM FILE vm-next then STREAM FILE vm-repeat |
| 10:31.07 | Zogot | i only hear repeat, but the asterisk console says it played both |
| 10:32.30 | Zogot | oh no, ignore me |
| 10:35.38 | phpboy | Zogot: I cannot get it to pass audio to my browser, if I do an rtp debug I see it sending packets to my local IP but I don't hear anything... got any tips? |
| 10:36.20 | phpboy | my browser is registered and I see it attempt to make the call on console |
| 10:37.34 | *** join/#asterisk italorossi (~Adium@177.193.104.232) |
| 10:39.13 | phpboy | it also takes a very long time to setup a call |
| 10:41.17 | Zogot | phpboy: you got the certificates? |
| 10:42.41 | phpboy | I do |
| 10:42.52 | phpboy | I generated them using the script in contrib/scripts |
| 10:43.13 | phpboy | I've disabled stun and ICE as I'll only be using this on a local network |
| 10:43.27 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 10:43.43 | phpboy | but for some reason when I disabled ice, it starts sending RTP packets to my gateway |
| 10:45.26 | Zogot | phpboy: this is my config https://gist.github.com/zogot/8e15829479940ad21ca9 |
| 10:45.44 | *** join/#asterisk Dratek (~mmoucka@kremouj.lit.rev-fortech.cz) |
| 10:46.44 | phpboy | you running on lan or public ips? |
| 10:46.53 | Dratek | hi guys. I have a question. If try to reload pjsip when my endpoints are registered Qualify works but if I unregister and register phone again it stays is Unavail status and qualify doesn't start. Any suggestions please? |
| 10:47.23 | phpboy | nat is disabled |
| 10:47.24 | phpboy | :\ |
| 10:47.53 | Zogot | phpboy: weve tested only from lan |
| 10:49.24 | phpboy | goodness knows why this thing refuses to use the lan IP for my client |
| 10:49.34 | phpboy | it's using the public IP for some bizar reason |
| 10:50.21 | Zogot | phpboy: what library are you using from the browser? |
| 10:51.30 | phpboy | sipml5 |
| 10:51.59 | Zogot | phpboy: we had issues with that one, we went with sipjs |
| 10:53.21 | phpboy | let me give that a shot |
| 10:53.23 | phpboy | thanks man |
| 10:54.25 | Zogot | phpboy: if you need a hand with configuration, give me a buzz |
| 11:05.43 | *** join/#asterisk Aamit (~Amit@103.54.136.32) |
| 11:06.10 | *** join/#asterisk _abc_ (~user@unaffiliated/ccbbaa) |
| 11:06.47 | _abc_ | Hello. Anyone active here at this time or should I come back later? |
| 11:07.35 | _abc_ | I would like to know if anyone has a Nortel/Avaya 2033 conference phone (the UFO shaped one) working with any version of asterisk using chan_unistim ? Or any other channel (I think these phones also do SIP with different firmware)? |
| 11:08.52 | _abc_ | http://community.polycom.com/t5/VoIP/Change-conference-phone-2033-from-UNIStim-to-H323-or-SIP/td-p/47057 I retract that |
| 11:18.02 | phpboy | Zogot: do you mind pasting your javascript config? |
| 11:18.18 | phpboy | if I look at debug in the browser it's trying to force ICE |
| 11:18.27 | phpboy | which I'd prefer it avoid if posible |
| 11:23.24 | *** join/#asterisk ThomasKeller (~Thomas@vmx.ethz.ch) |
| 11:27.25 | Zogot | phpboy: not sure how readable it'll be to you |
| 11:27.27 | Zogot | 1 sec |
| 11:27.44 | Zogot | phpboy: https://gist.github.com/zogot/c8620bb96fb09a1fd408 |
| 11:48.18 | *** part/#asterisk snadge (~snadge@unaffiliated/snadge) |
| 11:53.55 | *** join/#asterisk Parham (~chatzilla@46.209.201.131) |
| 11:54.07 | *** join/#asterisk vinrock (~vin@unaffiliated/vinrock) |
| 11:54.22 | *** part/#asterisk eschmidbauer (~eschmidba@unaffiliated/eschmidbauer) |
| 11:54.22 | Parham | Hi. How can I debug why I get this message? "Using SIP RTP CoS mark 5" |
| 11:56.09 | Parham | I get this message when trying to dial an extension. 'dialplan reload' works fine, and 'core reload' doesn't cause any errors. |
| 11:56.51 | *** join/#asterisk italorossi (~Adium@187.60.66.11) |
| 11:57.03 | *** join/#asterisk slav3_sergal (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 12:15.47 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:16.46 | *** join/#asterisk nny (601eebc0@gateway/web/freenode/ip.96.30.235.192) |
| 12:17.45 | WIMPy | Parham: You get that message when calling out via SIP and you have verbose set to 2 or higher. |
| 12:18.33 | Parham | WIMPy: Ah. That explains it. So, why am I not hearing the sound that the specified extension is supposed to play? I also don't get the customary reporting of the stack. |
| 12:20.05 | nny | Is dahdi-complete(ver)/linux/drivers/staging/echo the proper dir for oslec? Guides say cp -fr /usr/src/dahdi-linux-<ver> /usr/src/dahdi-complete(ver)/drivers/staging/echo but the drivers dir already exists |
| 12:20.14 | nny | er exists in linux |
| 12:20.31 | nny | ignore the cp -fr |
| 12:21.16 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 12:23.05 | WIMPy | wdoekes: By that kind of definition, I guess you could call the whole Asterisk a big bug. |
| 12:23.17 | simplydrew | Hey guys. Looking for methods to do monitoring of sip peers (end user phones and trunks) within asterisk to shoot me a simple email when something goes down. Trying to get ahead of client issues before they panic, when disaster strikes. Anyone have a way theyâre doing this now that they like? |
| 12:24.05 | WIMPy | simplydrew: Listen on AMI. |
| 12:25.29 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 12:26.12 | simplydrew | WIMPy: Ah, didnât think about utilizing the AMI. Not really a programmer, so was trying to see if there was something out there prebuilt to install, etc. |
| 12:26.57 | WIMPy | Well, it's pretty simple to listen for the peerstate events. |
| 12:28.42 | [TK]D-Fender | simplydrew: There are existing plugins for monit, etc |
| 12:28.54 | [TK]D-Fender | simplydrew: And the other big one |
| 12:29.06 | [TK]D-Fender | simplydrew: Don't remember the name because I've never touched them |
| 12:29.49 | simplydrew | [TK]D-Fender: Nice. Iâll have to take a look |
| 12:40.36 | nny | ok OSLEC is live, *sacrifices chicken* |
| 12:41.41 | *** join/#asterisk bkruse (~Adium@24.96.180.227) |
| 12:43.25 | nny | simplydrew: found this shell sript http://pastebin.com/eWHY9y0N |
| 12:44.05 | [TK]D-Fender | https://www.google.ca/#q=asterisk+monitoring+scripts |
| 12:44.11 | [TK]D-Fender | Nagios was the one I was thinking of. |
| 12:45.10 | simplydrew | nny: Very cool. I think thatâs what Iâm looking for. Probably will crontab that for every half hour or so |
| 12:45.39 | _abc_ | I would like to know if anyone has a Nortel/Avaya 2033 conference phone (the UFO shaped one) working with any version of asterisk using chan_unistim ? |
| 12:46.40 | nny | simplydrew: nice! |
| 12:46.59 | nny | ok I resolved my EC fun. Out for now |
| 12:57.46 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 12:59.39 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 12:59.45 | *** join/#asterisk generalhan (~tester@about/windows/staff/generalhan) |
| 13:04.55 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 13:07.09 | *** join/#asterisk blooregard (~francine@50.141.76.146) |
| 13:07.21 | blooregard | hi everyone |
| 13:10.47 | blooregard | I have a question regarding AMI vs ARI. I wrote an autodialer in perl several years ago before ARI was available that made extensive use of AMI to both send commands to asterisk an receive status messages to be logged back to the database. This dialer worked flawlessly the was it was written, however stupid me has managed to lose all of the code and I now have to rewrite the app. Would I be better off just sticking with what worked or should I po |
| 13:10.48 | blooregard | rt to ARI. I'll admit, I haven't really dug into the ARI details much at all. |
| 13:12.10 | [TK]D-Fender | AFAIK ARI is for creating apps used in actual call flow, not as a back-end manager |
| 13:12.16 | [TK]D-Fender | I'd read up on it in the WIKI |
| 13:14.17 | blooregard | That's kind of what I gathered. The dialer I wrote pulled numbers from a MySQL database, did an AMI originate, put the call in the correct extension in the dialplan then logged back the call status details to the database for the corresponding number dialed. |
| 13:15.41 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 13:16.53 | blooregard | I still can't believe i lost all that code. |
| 13:19.00 | blooregard | I'd like to be able to commercialize the autodialer somehow. |
| 13:25.23 | rl1 | 220 active channels |
| 13:25.23 | rl1 | 110 active calls |
| 13:25.24 | rl1 | yay |
| 13:27.17 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 13:27.25 | phpboy | [TK]D-Fender: have you ever played with webrtc? |
| 13:27.31 | [TK]D-Fender | nope |
| 13:29.46 | phpboy | :< |
| 13:39.02 | *** join/#asterisk vinrock (~vin@unaffiliated/vinrock) |
| 13:39.42 | *** join/#asterisk sruffell (sruffell@asterisk/the-kernel-guy/sruffell) |
| 13:39.42 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 13:42.09 | *** join/#asterisk monsterco (~monsterco@70.51.53.82) |
| 13:46.49 | *** join/#asterisk cervajs2 (~cervenka@gatekeeper.bm.ipex.cz) |
| 13:47.27 | cervajs2 | hi, can you recommend node.js module for fast AGI? there is maaaany options |
| 13:48.24 | *** join/#asterisk wonderworld (~ww@89.204.139.151) |
| 13:56.49 | *** join/#asterisk sparetire_ (~sparetire@unaffiliated/sparetire) |
| 14:01.12 | *** join/#asterisk bulkorok (~Adium@89.245.151.228) |
| 14:08.29 | *** join/#asterisk kharwell (kharwell@nat/digium/x-hipdoeeaiwwbuzew) |
| 14:09.09 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 14:11.38 | *** join/#asterisk marlinc_ (~marlinc@51.254.145.186) |
| 14:16.00 | *** part/#asterisk cervajs2 (~cervenka@gatekeeper.bm.ipex.cz) |
| 14:19.08 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-iqhvsmmsbteaudwp) |
| 14:26.51 | *** join/#asterisk marlinc (~marlinc@51.254.145.186) |
| 14:32.25 | *** join/#asterisk marlinc (~marlinc@51.254.145.184) |
| 14:33.48 | *** join/#asterisk u0m3 (~u0m3@89.120.204.99) |
| 14:39.34 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 14:41.21 | *** join/#asterisk bulkorok (~Adium@89.245.151.228) |
| 14:41.29 | *** join/#asterisk marlinc (~marlinc@51.254.145.184) |
| 14:48.48 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-gwozdsvcuvxuzrho) |
| 15:08.53 | *** join/#asterisk vader- (~Adium@50.232.174.194) |
| 15:08.54 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 15:14.07 | *** join/#asterisk Harm133 (2e2caf3e@gateway/web/freenode/ip.46.44.175.62) |
| 15:14.48 | Harm133 | Hello, I was wondering if there is any way to show in the logging which party disconnects the call, as in hangs up ? Im using freepbx |
| 15:15.29 | Zogot | Harm133: you'll have to go to the freepbx channel I think |
| 15:17.32 | Harm133 | Is such a feature even available within asterisk ? |
| 15:17.48 | Harm133 | ( I've already asked in the freepbx channel now, thanks ) |
| 15:17.56 | Zogot | Harm133: you can do things on the h context, which only triggers when the other end hangs up |
| 15:18.08 | Zogot | iirc |
| 15:18.13 | Zogot | actually im not sure now lol |
| 15:18.41 | *** join/#asterisk u0m3 (~u0m3@89.120.204.99) |
| 15:19.10 | *** join/#asterisk wonderworld (~ww@ip-84-119-184-180.unity-media.net) |
| 15:20.17 | [TK]D-Fender | Harm133: Answered in #freepbx |
| 15:37.29 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 15:41.17 | *** join/#asterisk superscrat (asanders@nat/digium/x-vtqsocahdqtkcagc) |
| 15:41.44 | wdoekes | WIMPy: I'd say that many things just work in a sane fashion. but cdr's don't. and you don't find out, until it's too late and you've already tied your billing to it |
| 15:45.07 | *** join/#asterisk marlinc (~marlinc@51.254.145.184) |
| 15:49.30 | *** join/#asterisk DragonAzul (~DragonAzu@187.208.12.95) |
| 15:52.30 | *** join/#asterisk eofster (~eofster@213.61.153.26) |
| 15:59.46 | WIMPy | wdoekes: CDRs were not a part where I had to seriousely lower my expectations. |
| 16:05.06 | _abc_ | Has anyone got a unistim Avaya/Nortel 2033 going with any version of asterisk please? |
| 16:06.23 | [TK]D-Fender | _abc_: Expect virtually no answqer here on it |
| 16:06.31 | [TK]D-Fender | Almost nobody uses UNISTIM. |
| 16:06.39 | [TK]D-Fender | Fewer still have that specific model |
| 16:06.54 | [TK]D-Fender | And then out of those who do ... how many would actually be sitting around in IRC? |
| 16:07.15 | _abc_ | [TK]D-Fender: One hopes, one repeats (infrequently) for a while. |
| 16:07.22 | [TK]D-Fender | Use the mailing list. |
| 16:07.46 | [TK]D-Fender | Do you have other UNISTIM phone working? |
| 16:07.51 | *** join/#asterisk monsterco (~monsterco@63.250.127.244) |
| 16:07.55 | _abc_ | [TK]D-Fender: Because *I* have such and asterisk crashes the phone (reboots) upon keypress on the phone ... actually when * sends the setup voice channel commands. |
| 16:08.11 | _abc_ | Yes I have plenty of unistim phones working and I hacked the channel a bit to make them work better. |
| 16:08.29 | _abc_ | So when I'll get bored with asking I'll go back to making it work, when I have time ;) |
| 16:08.43 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 16:09.00 | [TK]D-Fender | If you've already haved your setup then you should taking this to #asterisk-dev as you're not running the distributed code. |
| 16:09.27 | *** join/#asterisk DivideBy0 (~DivideBy0@unaffiliated/divideby0x0) |
| 16:09.40 | _abc_ | Yes, probably. Thanks for the tip. Meanwhile I'd like to catch a user, any user, on any version, having ever seen it work. I hope you can see how that can save lots of time. |
| 16:09.45 | _abc_ | +wasted |
| 16:10.11 | _abc_ | So I'll lurk a bit more here and infrequently ask the pesky question. If you don't mind. |
| 16:10.38 | _abc_ | Is Hans Cedric still actively contributing to chan_unistim? I talked to him some years ago about what I did. |
| 16:10.46 | [TK]D-Fender | Your odds are in "lightning strike" territory here. Mailing list is a better option |
| 16:10.55 | _abc_ | Yes, eventually. |
| 16:11.01 | _abc_ | feels lucky ;) |
| 16:13.36 | *** join/#asterisk marlinc (~marlinc@51.254.145.184) |
| 16:30.13 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-ljmsxjajpculkxfl) |
| 16:30.13 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 16:30.14 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 16:33.01 | *** join/#asterisk marlinc (~marlinc@51.254.145.184) |
| 16:38.49 | *** join/#asterisk igcewieling1 (~ewieling@ip98-170-211-145.pn.at.cox.net) |
| 17:07.04 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 17:07.09 | *** join/#asterisk Voyage (~user1@unaffiliated/voyage) |
| 17:07.55 | Voyage | in exten => 123,n,WaitForRing(5), 5 is the number of seconds to wait for a ring bell. whats 123 for? and whats n? I cant find docs for this. |
| 17:08.19 | malcolmd | 123 is the extension to match on |
| 17:08.23 | Voyage | some use 'same => " in stead of exten too. why |
| 17:08.25 | malcolmd | n is the priority |
| 17:08.26 | Voyage | malcolmd, hm |
| 17:08.32 | malcolmd | "same" is a shortcut |
| 17:08.38 | malcolmd | so you don't have to type the extension over and over |
| 17:08.39 | [TK]D-Fender | 123 = exten |
| 17:08.40 | Voyage | shortcut for exten? |
| 17:08.48 | [TK]D-Fender | And you said you understood this yesterday..... |
| 17:08.52 | [TK]D-Fender | is IT the exten |
| 17:09.01 | [TK]D-Fender | READ THE DAMN BOOK |
| 17:09.02 | igcewieling1 | wow. just wow. |
| 17:09.11 | [TK]D-Fender | IT is THE EXTEN* |
| 17:09.33 | Voyage | is the details in the book or the asterisk wiki/docs/ |
| 17:09.42 | [TK]D-Fender | BOOK <- |
| 17:09.46 | [TK]D-Fender | read it |
| 17:09.52 | Voyage | not asterisk docs? |
| 17:09.54 | igcewieling1 | The wiki is pretty useless for lots of stuff. |
| 17:09.57 | [TK]D-Fender | It's free. It's online. DO IT |
| 17:09.58 | Voyage | hm |
| 17:10.10 | Voyage | the orielly one I guess. ok |
| 17:10.17 | igcewieling1 | ~book |
| 17:10.21 | 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 |
| 17:11.04 | malcolmd | contexts, extensions, and priorities - https://wiki.asterisk.org/wiki/display/AST/Contexts%2C+Extensions%2C+and+Priorities |
| 17:11.06 | [TK]D-Fender | 100 times now. dialplan = extensions.conf and is well documented. Read it. No excuses, no delays. |
| 17:11.14 | igcewieling1 | [TK]D-Fender: you should answer every question with the relevant page number from the book. 8-) |
| 17:11.22 | lvlinux | lol |
| 17:11.49 | Voyage | :| |
| 17:11.56 | [TK]D-Fender | No, I seriously shouldn't have to |
| 17:12.28 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 17:12.42 | Voyage | <PROTECTED> |
| 17:12.53 | [TK]D-Fender | You didn't read. |
| 17:13.03 | [TK]D-Fender | and I also told you what each part of that line meant here |
| 17:13.06 | Voyage | [TK]D-Fender, Ok, yes, i have no cert to show |
| 17:13.10 | [TK]D-Fender | and you then you tell us you DO understand |
| 17:13.11 | lvlinux | Voyage: really, do read the book. It's great and it will get you started. Then we can help you with your confusing things. |
| 17:13.16 | [TK]D-Fender | And then you ask again and clearly don't |
| 17:13.19 | Voyage | lvlinux, ok! |
| 17:13.21 | igcewieling1 | Voyage: asking "what exten means" shows a fundamenta lack of knowledge. |
| 17:13.44 | Voyage | igcewieling1, I didnt found the terms in wiki :) |
| 17:13.46 | Voyage | ok. |
| 17:13.49 | Voyage | steps back |
| 17:13.59 | igcewieling1 | Voyage: no, you would not. That is how fundamental it is. |
| 17:14.18 | [TK]D-Fender | Dialplan basics. Extensions.conf. Can't find it? REALLY? You're going to try to tell us after all this you can't find it? |
| 17:14.24 | *** part/#asterisk Voyage (~user1@unaffiliated/voyage) |
| 17:14.53 | malcolmd | the usage of "same" as a replacement for the named extension is found on the wiki page i referenced. is the page not clear about what extensions are? |
| 17:15.07 | malcolmd | "Within each context, we can define one or more extensions. An extension is simply a named set of actions. Asterisk will perform each action, in sequence, when that extension number is dialed. " |
| 17:15.39 | [TK]D-Fender | ....and he's gone |
| 17:16.11 | lvlinux | hopefully going to read :-) |
| 17:17.12 | [TK]D-Fender | I'm not holding my breath |
| 17:17.36 | lvlinux | hehe |
| 17:24.08 | *** join/#asterisk Voyage (~user1@unaffiliated/voyage) |
| 17:24.16 | Voyage | maybe this pdf is more handy http://asterisk-service.com/downloads/Asterisk-%20The%20Definitive%20Guide,%204th%20Edition.pdf |
| 17:25.25 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 17:25.41 | lvlinux | Voyage: that works |
| 17:26.55 | Voyage | at least I will have a copy on my desktop |
| 17:27.04 | Voyage | and I can scroll fast too |
| 17:27.44 | WIMPy | thought that browsers scroll faster than pdf viewers. |
| 17:33.03 | *** join/#asterisk DivideBy0 (~DivideBy0@unaffiliated/divideby0x0) |
| 17:39.25 | _abc_ | browsers scroll faster than pdf.js crap viewers written in js and foisted upon unsuspecting firefox users |
| 17:39.31 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 17:39.48 | _abc_ | otherwise, pdf viewers scroll faster. |
| 17:40.39 | _abc_ | recommends sumatrapdf on windows - no page rotation and no search but very fast |
| 17:40.52 | *** join/#asterisk bkruse (~Adium@162.17.78.98) |
| 17:41.37 | [TK]D-Fender | Voyage: That IS the book we've been telling you a hundred times to go read |
| 17:42.57 | Voyage | I read.. the things I wanted to dive int |
| 17:45.01 | [TK]D-Fender | You need to master the dialplan. |
| 17:45.23 | [TK]D-Fender | and represents 95% of configuring * |
| 17:49.32 | *** join/#asterisk mac_ified (~mac_ified@67-9-150-210.res.bhn.net) |
| 17:50.11 | DivideBy0 | down with dialplan! |
| 17:59.39 | WIMPy | libasterisk |
| 17:59.51 | WIMPy | Was that Asterisk 20? |
| 18:07.01 | Voyage | [TK]D-Fender, yup |
| 18:07.08 | DivideBy0 | long live dialplan! |
| 18:08.14 | Voyage | can anyone recommend a 'one channel unlimited package' plan for fixed per month cost (US calling)? |
| 18:09.10 | [TK]D-Fender | What are you expecting to do with this? |
| 18:09.39 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 18:09.53 | Voyage | [TK]D-Fender, call, of course |
| 18:10.07 | lvlinux | flowroute has one for $25 I think. |
| 18:10.08 | [TK]D-Fender | How much are you actually INTENDING to call? |
| 18:10.11 | Voyage | we dont have much calling load for now. so one chanel is fine |
| 18:10.17 | Voyage | lvlinux, no, thats for inbound calls only |
| 18:10.21 | Voyage | lvlinux, the vPRI |
| 18:10.27 | lvlinux | ah, yes that's right. |
| 18:10.41 | Voyage | [TK]D-Fender, not very hight. but we are comfortable with per month fixed |
| 18:10.47 | Voyage | high |
| 18:11.59 | [TK]D-Fender | $25/month is the same as paying for 2500 minutes (41.6) hours a month @ $0.01/min regardless of whether you use it or not. |
| 18:12.44 | Voyage | [TK]D-Fender, right, still its a psychic / comfort thing |
| 18:16.27 | Voyage | http://www.sipstation.com is giving unlimited plan for $25 |
| 18:19.09 | WIMPy | If unlimited would make sense for the customer, it probably wouldn't exist. |
| 18:19.33 | [TK]D-Fender | that isn't "unlimited". |
| 18:19.42 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 18:19.48 | WIMPy | Trouble is that it's hard to find any plan these days that doesn't include unlimited national PSTN :-( |
| 18:19.49 | *** join/#asterisk u0m3 (~u0m3@89.120.204.99) |
| 18:19.49 | [TK]D-Fender | Like all there is a soft-limit of 3000min , plus other T&C. |
| 18:19.54 | *** part/#asterisk _abc_ (~user@unaffiliated/ccbbaa) |
| 18:20.07 | WIMPy | Unlimited is never unlimited, but it might be called so. |
| 18:20.32 | lvlinux | google voice! lol... |
| 18:21.06 | *** join/#asterisk marlinc (~marlinc@bouncer.cvo.technology) |
| 18:21.43 | [TK]D-Fender | "Additionally, Sangoma reserves the right to in any combination (i) immediately terminate the Service, (ii) charge a minimum inappropriate use fee of $500.00 and/or charge $0.05 per minute for all calls made during such periods of prohibited use (plus applicable toll free and international charges), whichever is higher, to Customerâs payment method of record, and (iii) all applicable... |
| 18:21.44 | [TK]D-Fender | ...termination fees described in these Terms and Conditions." |
| 18:22.29 | WIMPy | Is that legal? |
| 18:23.08 | igcewieling1 | When has legality ever applied to contracts? 8-| |
| 18:23.19 | WIMPy | Good point. |
| 18:23.25 | [TK]D-Fender | Of course it is. That's in the T&C. It's a contract. You accept it in order to become a customer. |
| 18:23.50 | WIMPy | Such T&C would be void here. |
| 18:24.45 | WIMPy | Which means all of th4e T&C unless being taken care of. |
| 18:25.55 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 18:27.39 | phpboy | WIMPy: have you ever implemented webrtc in asterisk? |
| 18:28.14 | WIMPy | Nope |
| 18:28.15 | [TK]D-Fender | phpboy: Rather than asking every user one by one how about just asking your actual question? |
| 18:28.36 | WIMPy | Like in |
| 18:28.38 | WIMPy | ~ask |
| 18:28.42 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 18:28.42 | phpboy | [TK]D-Fender: I've resorted to asking specific people :P |
| 18:28.46 | WIMPy | ? |
| 18:28.46 | [TK]D-Fender | Stop |
| 18:28.51 | phpboy | ok |
| 18:29.09 | WIMPy | Anyway... I'm off for the evening. |
| 18:29.11 | [TK]D-Fender | Just get started and when you run into snag ask sepcific question and show where you're at |
| 18:30.29 | phpboy | so I've implemented webrtc and tried two 'clients' sipxl5 and sip.js.... I cannot get it to pass audio for the life of me. the asterisk server is on the same network as the web clients... LAN only... so my first question is do I need to use stun/ice and how can I pass audio directly between asterisk and the web clients |
| 18:30.51 | phpboy | audio issues are quite common online, but none of the threads have concrete answers |
| 18:33.33 | robmal | Can u haz pcap? |
| 18:35.36 | phpboy | right now that's a bit of an issue |
| 18:35.47 | phpboy | was hoping to cover some of the theory |
| 18:35.57 | [TK]D-Fender | Theory won't prove your problem. |
| 18:36.01 | robmal | In theory something is wrong. |
| 18:36.14 | phpboy | my biggest question is, can this work without ICE/STUN |
| 18:36.18 | robmal | Yes. |
| 18:43.18 | Voyage | WIMPy, do you know any so called unlimited one? |
| 18:44.26 | [TK]D-Fender | none are likely to be "unlimited |
| 18:44.29 | [TK]D-Fender | You missed the point |
| 18:44.45 | [TK]D-Fender | All the companies that ever offered these WITHOUT restriction learned that they get abused |
| 18:44.50 | [TK]D-Fender | You are looking for a UNICORN |
| 18:51.19 | robmal | https://cassiecramer.files.wordpress.com/2011/01/hippo-to-unicorn.jpg |
| 18:52.08 | [TK]D-Fender | REAL UNICORNS HAVE CURVES! #allunicornsarebeautiful |
| 18:53.47 | robmal | http://s1.favim.com/orig/7/horse-magical-rainbow-rhino-unicorn-Favim.com-166598.jpg |
| 18:57.04 | *** join/#asterisk TriJetScud (~TriJetScu@van-app-svr.ad.v10networks.ca) |
| 18:57.04 | *** join/#asterisk nanoha-sama (~nanoha-sa@van-app-svr.ad.v10networks.ca) |
| 18:58.15 | *** join/#asterisk davidbowlby (~textual@cpe-71-79-82-152.columbus.res.rr.com) |
| 19:00.58 | davidbowlby | I'm running some phones behind nat with nat=yes qualify=yes and was curious about phones I take offline. I see the system trying to send options messages (which I understand why) for long intervals after the phone has been offline. Is there a timeout option where asterisk will give up sending OPTION messages on the last known NAT address? |
| 19:02.26 | phpboy | ok, so I think I found the problem kind of... if I do a sip set debug... the from in my sip headers says 1234@192.168.8.70 (192.168.8.70 being my astirisk server) but my client is 192.168.7.248... what could cause this besides the client? |
| 19:02.45 | davidbowlby | phpboy, nat? |
| 19:03.26 | phpboy | no nat on this network, it's routed... not even a firewall between the two (Layer 3 switch) |
| 19:04.05 | phpboy | SIP read from WS: says my client IP by the FROM in the SIP header clearly says the asterisk servers IP |
| 19:04.07 | davidbowlby | phpboy, well doesn't it go to 1234@<asterisk IP> then asterisk sends it to 1234@<client IP> |
| 19:04.35 | phpboy | that's the thing, I don't know |
| 19:04.43 | davidbowlby | phpboy, I just got in the channel, so didn't get the background on what you're going through. |
| 19:04.53 | davidbowlby | phpboy, but that is normally what you see |
| 19:04.56 | phpboy | WEBRTC client |
| 19:05.02 | davidbowlby | phpboy, ah |
| 19:06.01 | *** part/#asterisk Voyage (~user1@unaffiliated/voyage) |
| 19:06.35 | phpboy | in all the sip headers... always from: 1234@192.168.8.70 and to: 1234@192.168.8.70 ... so it's looping I think and not sure where to fix it |
| 19:06.59 | [TK]D-Fender | Don't just say "headers". |
| 19:07.03 | [TK]D-Fender | It matters WHICH ONES |
| 19:08.34 | phpboy | I'm looking through all of them, not much change... |
| 19:09.04 | [TK]D-Fender | "all of them" tells us nothing you know... |
| 19:09.28 | phpboy | ok, which should I be looking for specifically? or is a pastebin going to be better? |
| 19:09.42 | [TK]D-Fender | ALWAYS better |
| 19:09.52 | robmal | You're looking for INVITE, it specifies where the rtp stream goes to. |
| 19:10.12 | robmal | Also: 1234@192.168.8.70 is good. |
| 19:10.32 | phpboy | please hold, changing hosts... moving myself into the same block as the asterisk server to iron out any NAT realated issues/questions |
| 19:14.13 | phpboy | http://pastebin.ca/3221819 <----- * is 192.168.8.70, my webrtc client/browser is 192.168.8.160 and 192.168.8.100 is another server on my network trying. this one can be ignored |
| 19:15.35 | robmal | o=mozilla...THIS_IS_SDPARTA-41.0.2 4294967295 0 IN IP4 192.0.2.75 |
| 19:15.38 | robmal | This is wrong. |
| 19:17.01 | phpboy | I have no idea what that is or means? |
| 19:17.14 | robmal | Where does this 192.0.2.75 get from? |
| 19:17.26 | phpboy | I have no idea |
| 19:17.32 | phpboy | let me check the sip.js source |
| 19:18.40 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 19:18.41 | davidbowlby | sounds natty ;) |
| 19:18.44 | phpboy | I have no idea where that comes from |
| 19:19.02 | davidbowlby | phpboy, I bet your network guy does |
| 19:19.03 | phpboy | it does, but I'm on the same subnet/network |
| 19:19.09 | phpboy | I am my network guy |
| 19:19.14 | davidbowlby | phpboy lol |
| 19:19.18 | robmal | Go get a mirror. |
| 19:19.24 | phpboy | I am physically on the same network |
| 19:19.34 | davidbowlby | lol a mirror, yes and be harsh with him |
| 19:19.49 | davidbowlby | phpboy, maybe your software has that address in a config file or option? |
| 19:19.50 | phpboy | I will give him uphill |
| 19:19.59 | davidbowlby | phpboy, the client |
| 19:20.17 | phpboy | well, I did a quite search in the .js file, nothing uses that IP |
| 19:20.30 | phpboy | but besides that, what else looks strange? |
| 19:20.49 | davidbowlby | your register fails |
| 19:20.57 | davidbowlby | phpboy, do you have it set to dynamic? |
| 19:21.09 | davidbowlby | phpboy, is the secret correct? |
| 19:22.05 | davidbowlby | phpboy, actually one fails, then works.. |
| 19:22.23 | phpboy | host=dynamic yes |
| 19:23.05 | phpboy | the reason why you may be seeing that is because I do a refresh of the page which probably causes deregister and reregister |
| 19:23.16 | davidbowlby | ah |
| 19:23.39 | phpboy | the secret etc must be right because I can see it 'attempt' the call by running through the dial plan |
| 19:23.59 | robmal | Find that ip. |
| 19:23.59 | davidbowlby | phpboy, yeah, it would have rejected the invite |
| 19:24.01 | phpboy | what's weird is it stops at exten => 1234,1,Answer() |
| 19:24.24 | phpboy | the next step is to play a recording for testing |
| 19:24.27 | davidbowlby | I agree with robmal, it's trying to do RTP on that port |
| 19:24.30 | *** join/#asterisk Tim_Toady (~fuzzy@snf-33276.vm.okeanos.grnet.gr) |
| 19:24.35 | davidbowlby | phpboy, how many interfaces are on this host? |
| 19:24.40 | phpboy | 1 |
| 19:24.40 | davidbowlby | and your web server |
| 19:25.07 | phpboy | asterisk/web = 1 interface |
| 19:25.44 | phpboy | something that's worth mentioning is this was installed on a CentOS 7 server for what that's worth |
| 19:25.47 | robmal | Holy shit, google is fast now. |
| 19:25.57 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 19:26.07 | robmal | Your pastebin is on 9th position for: "192.0.2.75" sip |
| 19:26.19 | phpboy | damn |
| 19:26.21 | phpboy | that is fast |
| 19:26.23 | davidbowlby | lol |
| 19:26.44 | phpboy | 192.0.2.75 is in public address space |
| 19:26.54 | davidbowlby | with nosql, it's the old stuff that's hard to find ;) |
| 19:27.37 | davidbowlby | phpboy, it looks like it is coming from your browser, do you have a proxy set up? |
| 19:27.57 | davidbowlby | phpboy, web proxy might do this |
| 19:28.06 | phpboy | let me double check |
| 19:28.15 | phpboy | it's an IP range in the states, I'm in South Africa |
| 19:28.16 | davidbowlby | autoconfig burn maybe? |
| 19:28.20 | phpboy | so I doubt it |
| 19:28.47 | phpboy | definitely no proxy |
| 19:28.59 | davidbowlby | I don't think 192.0 is public |
| 19:29.04 | phpboy | it is |
| 19:29.13 | robmal | 192.168.0.0/16 is private. |
| 19:29.20 | phpboy | bingo |
| 19:29.30 | robmal | phpboy: tcpdump the whole thing for me. |
| 19:29.47 | phpboy | please hold |
| 19:29.57 | robmal | ... your call is important to us. |
| 19:31.08 | phpboy | tcpdump -i ens32 host 192.168.8.160 and not port ssh |
| 19:31.14 | phpboy | will that suffice? |
| 19:31.21 | phpboy | or should I through in -vv ? |
| 19:31.54 | robmal | -i any -s 65535 -w whatever port not ssh |
| 19:33.08 | phpboy | ah, you want the actual pcap |
| 19:33.12 | phpboy | how do I get it to you? |
| 19:33.42 | robmal | wetransfer |
| 19:35.27 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 19:36.16 | phpboy | let me put it on my webserver instead |
| 19:36.30 | robmal | But i like wetransfer :-( |
| 19:36.41 | phpboy | it seems complicated :\ |
| 19:36.50 | phpboy | paste me a link to upload it please |
| 19:37.13 | robmal | https://www.wetransfer.com/howitworks |
| 19:38.11 | robmal | Just enter some fake emails. |
| 19:38.16 | *** join/#asterisk SpeakerToMeat (~SpeakerTo@prgmr/customer/SpeakerToMeat) |
| 19:38.20 | SpeakerToMeat | Hello. |
| 19:38.23 | phpboy | http://www.phpboy.co.za/robmal.pcap |
| 19:38.30 | phpboy | wetransfer to follow... |
| 19:39.17 | robmal | Oh, you need to click the share icon and switch it to link instead of email |
| 19:39.59 | SpeakerToMeat | I come, as usual, to make a stupid question. I'm sorry... question if I have set up a sip trunk with encryption=yes and transport=tls, is there any way I can get/use a certificate file and config asterisk so it'll connect to the endpoint only if the certificate matches? |
| 19:40.01 | phpboy | http://we.tl/DvoGBFSYcP |
| 19:42.42 | robmal | o=mozilla...THIS_IS_SDPARTA-41.0.2 4294967295 0 IN IP4 192.0.2.59 |
| 19:42.54 | robmal | Uhm, something is seriously wrong ;-) |
| 19:45.07 | igcewieling1 | perhaps webRTC? |
| 19:45.10 | robmal | K, lets try the easy way ;-) |
| 19:45.14 | robmal | phpboy: Try chrome. |
| 19:45.42 | robmal | BTW, there was some magic website for sip debuging. |
| 19:46.55 | phpboy | yeah? |
| 19:47.35 | davidbowlby | goatse.cx? |
| 19:48.52 | igcewieling1 | SpeakerToMeat: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling |
| 19:50.28 | robmal | Ye! |
| 19:50.30 | robmal | http://mypcap.com/ |
| 19:51.53 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 19:52.24 | SpeakerToMeat | igcewieling1: Thank you |
| 19:56.05 | phpboy | robmal: http://pastebin.ca/3221903 <---- sip debug with chrome |
| 19:57.07 | robmal | Even better ;-) |
| 19:57.26 | phpboy | still having same issue :( |
| 19:57.54 | robmal | Because the rtp stream is sent to 127.0.0.1 ;-) |
| 19:58.06 | phpboy | yes, that I did notice |
| 19:58.13 | robmal | I'd suggest starting over with the sipml5 config. |
| 19:58.13 | phpboy | why would that be though? |
| 19:58.29 | *** join/#asterisk GameGamer43 (sid5533@gateway/web/irccloud.com/x-edxqggopbwhoqrba) |
| 19:58.33 | phpboy | in this case I'm using sip.js... forgot its formal name |
| 19:58.55 | phpboy | I could even use the puplic sipml5 |
| 19:59.07 | phpboy | live demo |
| 20:00.27 | robmal | So check it. |
| 20:00.47 | *** join/#asterisk jeffspeff (~jeff.clay@12.49.160.131) |
| 20:01.45 | *** part/#asterisk igcewieling1 (~ewieling@ip98-170-211-145.pn.at.cox.net) |
| 20:03.19 | phpboy | it's late here |
| 20:03.22 | phpboy | gotta sleep |
| 20:03.25 | phpboy | thanks a lot for your help |
| 20:03.35 | phpboy | I hope tomorrow brings me fortune |
| 20:04.19 | robmal | Good luck. |
| 20:06.25 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 20:11.42 | *** join/#asterisk robink_ (~quassel@unaffilated/robink) |
| 20:19.35 | *** join/#asterisk wonderworld (~ww@ip-84-119-184-180.unity-media.net) |
| 20:22.02 | *** join/#asterisk generalhan (~tester@about/windows/staff/generalhan) |
| 20:23.09 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-172.cust.bezeqint.net) |
| 20:25.22 | *** join/#asterisk Oatmeal (~Suzeanne@75-103-145-152.ccrtc.com) |
| 20:30.40 | *** join/#asterisk camerin (hoax@elite.bshellz.net) |
| 20:31.47 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 22:35.03 | *** join/#asterisk infobot (ibot@69-58-76-73.ut.vivintwireless.net) |
| 22:35.03 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.6.0 (2015/10/09), 11.20.0 (2015/10/09), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 22:52.47 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 22:56.15 | *** join/#asterisk monsterco (~monsterco@70.51.53.82) |
| 23:26.27 | *** join/#asterisk superscrat (~asanders@173-21-89-217.client.mchsi.com) |
| 23:28.22 | *** part/#asterisk kharwell (kharwell@nat/digium/x-hipdoeeaiwwbuzew) |
| 23:39.30 | *** join/#asterisk saint_ (~saint_@c-71-59-68-185.hsd1.nj.comcast.net) |
| 23:40.14 | saint_ | hi all, how can i limit or delete the console history in asterisk 11 ? |
| 23:40.55 | WIMPy | Just delete the file. |
| 23:41.32 | saint_ | WIMPy which one is it ? |
| 23:41.50 | robmal | The big one. |
| 23:41.55 | WIMPy | The obvious one :-) |
| 23:42.46 | WIMPy | In case you don't see it: .asterisk_history |
| 23:46.56 | saint_ | damn me. |
| 23:47.00 | saint_ | that was obvious .. |
| 23:53.47 | saint_ | gotta go , thanks |
