| 00:00.02 | [TK]D-Fender | whre is the SIP DEBUG? |
| 00:00.07 | [TK]D-Fender | You went back to NOT LOOKING |
| 00:00.30 | mophed | omg now that sip debug works. |
| 00:00.44 | mophed | earlier it said that was not a command. |
| 00:00.56 | [TK]D-Fender | Yeah when I told yout he modules was NOT LOADED |
| 00:01.01 | [TK]D-Fender | And LOADED IT |
| 00:01.05 | mophed | oooooohhhhhh. |
| 00:01.20 | mophed | jesus. |
| 00:01.23 | [TK]D-Fender | and after doing so... YOU DID NOT ENABLE DEBUG |
| 00:01.41 | [TK]D-Fender | Today's magic word is : thorough |
| 00:02.50 | mophed | can i send you a case of beer? |
| 00:02.56 | mophed | http://pastebin.ca/3755290 |
| 00:04.13 | mophed | i see that * is again registered on the flowroute site |
| 00:04.51 | mophed | and lo and behold the xlite phone is connected to * |
| 00:13.56 | [TK]D-Fender | Retransmitting #1 (NAT) to 216.115.69.144:5060: |
| 00:13.57 | [TK]D-Fender | REGISTER sip:sip.flowroute.com SIP/2.0 |
| 00:14.06 | [TK]D-Fender | Flowroute is NOT behind nat and I told you to fix the peer |
| 00:14.13 | [TK]D-Fender | You have clearly NOT done so |
| 00:14.28 | [TK]D-Fender | Contact: <sip:s@104.0.251.34:5060> |
| 00:14.57 | [TK]D-Fender | Also you would be advised to tell them a BETTER contact so that you can match something mroe uniquely identifiable when they call you. |
| 00:15.16 | [TK]D-Fender | Like the DID that registering to them to have routed to you |
| 00:15.31 | mophed | i havent purchased a DID |
| 00:15.35 | [TK]D-Fender | register =? user:pass@host/PUTTHECONTACTHERE |
| 00:15.45 | [TK]D-Fender | So you are only using them for termination? |
| 00:15.54 | [TK]D-Fender | No inbound number? |
| 00:16.52 | mophed | i was just waiting to do it because none of this was working. |
| 00:17.15 | [TK]D-Fender | Is that a "yes"? |
| 00:17.43 | mophed | when i connected directly to them from xlite the user id would show up on my caller id. but i was trying to get this configured with * |
| 00:17.44 | [TK]D-Fender | If so it's a very complicated substitute for the 2-3 letter answer I was looking for |
| 00:18.55 | mophed | im sorry. i will buy a DID if i can ever get this thing working but right now yes just termination |
| 00:19.14 | [TK]D-Fender | Again could have just been 2-3 letters |
| 00:19.21 | [TK]D-Fender | So... go fix your peer |
| 00:19.30 | [TK]D-Fender | And keep the registration in mind when you get a DID to go with that |
| 00:32.50 | *** join/#asterisk _nexxus_ (~bwg@leon.generalamalgamated.com) |
| 00:48.02 | mophed | the [trunk] in sip.conf is how outside calls are made? |
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| 01:03.02 | [TK]D-Fender | that is an incomplete idea |
| 01:03.33 | [TK]D-Fender | Call comes FROM your phone, matches a PEER for it, hits the DIALPLAN, and then you Dial() out using a PEER to your provider |
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| 01:17.26 | mophed | http://pastebin.ca/3755316 |
| 01:17.52 | mophed | chan_sip.c:26259 handle_request_invite: Call from '201' (192.168.1.181:5061) to extension '19124444444' rejected because extension not found in context 'from-internal'. |
| 01:18.06 | mophed | shouldnt that be dialing out and not to an extension. its an 11 digit number |
| 01:19.18 | mophed | i feel like im so close |
| 01:20.16 | WIMPy | Everythin goes to an extension. You need to configure one. |
| 01:21.59 | mophed | i have one with [outgoing] |
| 01:22.27 | WIMPy | But that's not what you're using. |
| 01:22.36 | mophed | so shouldt it be trying to use the outgoing and not 'from-inernal' |
| 01:22.39 | WIMPy | It says you using "from-internal". |
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| 01:24.34 | mophed | OMG!!!!!!!!! IT WORKED I MADE A CALL! |
| 01:24.43 | mophed | thank you guys soooooo much! |
| 01:25.25 | mophed | i love you D-Fender! you are the greatest, most knowledgeable and patient human ever! |
| 01:26.10 | mophed | i just copied what i had in [outgoing] and put it under from-internal and it worked |
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| 01:35.28 | [TK]D-Fender | probably not the way.... |
| 01:35.35 | [TK]D-Fender | but glad there's progress |
| 01:35.49 | [TK]D-Fender | put it BACk (broken) so we can see what you had and if there is a better way |
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| 02:38.18 | mophed | http://pastebin.ca/3755333 extensions |
| 02:38.25 | mophed | http://pastebin.ca/3755334 broken debug |
| 02:38.35 | mophed | sorry i had to take a celebration walk |
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| 02:42.11 | mophed | when i dial an 11 digit number souldnt it be trying to use [outgoing] what makes it try the 'from-internal' |
| 02:42.37 | WIMPy | It uses the context you configured for the device. |
| 02:43.04 | WIMPy | It serches for the extension (e.g. an 11 digit number) within that context. |
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| 02:46.20 | mophed | which file is that configured in? |
| 02:46.49 | WIMPy | Where your device is configured. In this case sip.conf. |
| 02:52.31 | mophed | i dont have a DID and the outgoing call is coming up on the caller id as +1(201) is there a way to change that? |
| 02:53.06 | mophed | exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=xxxxxxxxxx) ? |
| 02:53.23 | asteriskmonkey | yea that should work |
| 02:53.46 | asteriskmonkey | could also be your upstream doing that if thats failing |
| 02:53.55 | WIMPy | That really depends on what your provider understands and allows. |
| 02:54.17 | mophed | yeah, having that in extensions just gets the busy tone as if its not dialing out at all |
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| 03:01.06 | mophed | their site states i can use P-Asserted-Identity, Remote-Party-ID or From: |
| 03:02.32 | WIMPy | Maybe it was the wrong format. |
| 03:02.44 | WIMPy | Or you messed up your extension. |
| 03:08.12 | *** join/#asterisk u0m3_ (~u0m3@86.127.133.199) |
| 03:13.01 | mophed | exten => _X.,n,SIPAddHeader(P-Asserted-Identity: <sip:0755123456@202.32.44.55>) |
| 03:13.04 | mophed | exten => _X.,n,SIPAddHeader(Privacy:off) |
| 03:13.37 | mophed | that also did not work. but if i make a new device in sip.conf with the number id like to show. that works |
| 03:17.22 | asteriskmonkey | k probably using diffent field, you can probably just set callerid all command to not have to guess :) |
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| 04:00.51 | [TK]D-Fender | mophed, "sendrpid=yes", "trustrpid=yes" <- for your peer |
| 04:01.00 | [TK]D-Fender | and just set the callerid |
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| 05:11.16 | mophed | http://asterisk-service.com/downloads/Asterisk-%20The%20Definitive%20Guide,%204th%20Edition.pdf |
| 05:11.37 | mophed | would this be a good book to read? or can you suggest something better i should look at? |
| 05:13.40 | [TK]D-Fender | ~book |
| 05:13.40 | 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 |
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| 05:39.52 | *** join/#asterisk DoYouKnow (~DoYouKnow@unaffiliated/doyouknow) |
| 05:39.54 | DoYouKnow | hi |
| 05:40.03 | DoYouKnow | is there any way to fix the core dump issue on ubuntu 16.04? |
| 05:40.14 | DoYouKnow | (without compiling from source) |
| 05:40.35 | drmessano | Is the core dump issue with their package? |
| 05:40.59 | drmessano | As in, is it known to be an issue specific to the package? |
| 05:42.12 | DoYouKnow | I don't know |
| 05:42.19 | drmessano | .... |
| 05:42.34 | drmessano | Well it's impossible to answer that without some detail |
| 05:42.58 | drmessano | "The core dump issue on Ubuntu 16.04" isn't a thing |
| 05:44.48 | DoYouKnow | <PROTECTED> |
| 05:44.49 | DoYouKnow | asterisk: ../src/pjsip-simple/evsub.c:415: pjsip_evsub_register_pkg: Assertion `mod_evsub.mod.id != -1' failed. |
| 05:44.53 | DoYouKnow | followed by core dumped |
| 05:45.15 | DoYouKnow | https://bugs.launchpad.net/ubuntu/+source/asterisk/+bug/1631101 |
| 05:45.16 | drmessano | Which version of Asterisk? |
| 05:45.30 | DoYouKnow | Asterisk 13.1.0~dfsg-1.1ubuntu4 |
| 05:47.40 | drmessano | Thats going to be incredibly hard to nail down without compiling a current version. I would imagine contacting the maintainer is your next step because there is no generic core dump on startup issue with pjsip enabled. |
| 05:50.00 | drmessano | Your best bet is to not use packages a Yeah. |
| 05:50.05 | drmessano | Anyway* |
| 05:50.17 | drmessano | Sorry, autocorrect fail |
| 05:50.43 | drmessano | Compiling Asterisk for source is easy and worth it |
| 05:50.47 | drmessano | From |
| 05:51.30 | craigify | hey everyone |
| 05:52.44 | craigify | DoYouKnow, Do you need ODBC? |
| 05:52.57 | craigify | reason why I ask, is if you have any need for realtime stuff |
| 05:53.18 | craigify | ubuntu 16.04 LTS changed up the mysql odbc packages |
| 05:54.54 | drmessano | I think he wants it to not crash from the get go |
| 05:56.05 | craigify | I know. I should have been clearer. compiling from source with an odbc requirement doesn't work because ubuntu doesn't have the mysql driver. it probably could be downloaded manually, but it adds in an annoying extra step |
| 05:56.24 | drmessano | Meh |
| 05:56.37 | drmessano | The ODBC driver in Ubuntu has been fucked for years |
| 05:56.47 | drmessano | Use the one from MariaDB |
| 05:56.57 | craigify | well it worked up until 16 |
| 05:57.03 | craigify | perhaps that is a good idea |
| 05:57.13 | drmessano | No, it mostly worked |
| 05:57.28 | drmessano | It's been the same driver for years and flawed |
| 05:57.48 | drmessano | Lots of problems with newer installs |
| 05:58.19 | drmessano | I've recommended the MAriaDB driver version for a couple years |
| 05:58.32 | drmessano | It just works |
| 06:01.41 | drmessano | The included driver from < 16.04 was from 2012 and built just before an important bug was fixed for an API change. The maintainer never updated it because it "worked". But I had lots of working installs that stopped working with minor MySQL updates. Total nightmare |
| 06:02.10 | drmessano | The MariaDB driver is great. |
| 06:04.07 | DoYouKnow | I wanted the mysql CDR updates, as something to experiment with |
| 06:04.28 | DoYouKnow | but I'm thinking I can just apt-get remove asterisk, then compile just asterisk from source |
| 06:07.49 | drmessano | Good idea |
| 06:28.56 | *** join/#asterisk Rasputin3711 (~Rasputin3@87.255.254.66) |
| 06:45.32 | *** join/#asterisk xous (~xous@unaffiliated/xous) |
| 06:48.41 | xous | anyone recommend a good usb over the ear wireless headset for a softclient? |
| 06:50.47 | Rasputin3711 | we use plantronics devices in our call center |
| 06:51.30 | xous | I've got a CS55 for my hard phone with a HL10 |
| 06:51.48 | xous | sound/range/life time have been excellent |
| 06:52.46 | xous | but it doesn't directly hook up to a computer. |
| 06:52.58 | xous | Do you know what model you use? |
| 06:53.43 | [TK]D-Fender | USB BT + BT headset |
| 06:53.47 | [TK]D-Fender | plantronics makes a ton |
| 06:53.54 | [TK]D-Fender | So does everybody |
| 06:54.28 | xous | I have a jawbone bluetooth device somewhere around here. never use it because I couldn't get it comfortable |
| 06:55.00 | [TK]D-Fender | Well "good" and "comfortable for you" aren't something we can match up |
| 06:55.05 | xous | yeah |
| 06:55.13 | [TK]D-Fender | Reviews will say "good", the rest is on you |
| 06:56.32 | Rasputin3711 | cs540 |
| 06:56.53 | Rasputin3711 | cs510 |
| 06:56.55 | [TK]D-Fender | Not USB |
| 06:57.11 | [TK]D-Fender | Please read the "big print" requirement |
| 06:57.25 | xous | yeah, suppose i could find cable to break out a RJ 9 to mic + speaker |
| 06:57.31 | xous | but I doubt it would sound very good |
| 06:57.31 | [TK]D-Fender | I just ordered 3 x CS540's for my office last week |
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| 07:02.12 | xous | I guess i'll call our vendor tomorrow |
| 07:03.31 | xous | Plantronics Voyager looks interesting |
| 07:05.17 | *** join/#asterisk cmendes0101 (~cmendes01@47-144-223-7.lsan.ca.frontiernet.net) |
| 07:08.54 | drmessano | You can't just "Break out" an RJ9 to mic and speaker and connect to a computer. |
| 07:09.32 | xous | yeah, there are some adapters iirc |
| 07:10.04 | drmessano | I'm sure there are cables. There are cables for anything if you Google. |
| 07:10.11 | drmessano | But that doesn't mean they work |
| 07:10.30 | drmessano | You would need an interface with a matching network |
| 07:12.18 | xous | I don't understanding "networK" in this context |
| 07:12.35 | drmessano | Plugging a ~600 ohm load into a speaker jack doesn't work all that well. The microphone wouldn't even work. |
| 07:13.06 | xous | ah. yeah, I understand there would likely need to be some active electronics to make it work |
| 07:13.21 | drmessano | Yeah, *expensive box*.. not just a cable |
| 07:13.34 | xous | hehe |
| 07:13.44 | drmessano | I wasnt being funny |
| 07:13.44 | xous | maybe I oversimplified it |
| 07:14.03 | xous | I wasn't planning on going that route anyway. |
| 07:15.13 | xous | a coworker bought just a "3.5m to 4P4C" cable and it didn't work too well for him :P |
| 07:15.29 | xous | wanted to use his headphones with his phone |
| 07:15.48 | drmessano | No, because anything you find like that is for a SPECIFIC piece of hardware that happened to use those connectors |
| 07:16.16 | drmessano | There is no direct interface between RJ9/RJ11 to line/mic level |
| 07:18.32 | drmessano | I use any off the shelf bluetooth headset and ebay bluetooth dongles |
| 07:18.42 | drmessano | or built-in bluetooth |
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| 07:19.07 | drmessano | A $6 ebay Bluetooth dongle and a plantronics headset works fine |
| 07:19.48 | xous | yeah |
| 07:20.00 | xous | I have a few usb bluetooth receivers |
| 07:20.19 | xous | I'll see if I can expends one for "testing" |
| 07:20.35 | drmessano | $6? |
| 07:20.37 | xous | I want to see if I can move our receptionists to a soft client without anyone noticing. |
| 07:20.43 | xous | no idea what I paid |
| 07:20.48 | xous | but I already got 'em |
| 07:20.48 | *** join/#asterisk Tiffon (~name@unaffiliated/tiff0n) |
| 07:21.17 | drmessano | Ok, you said something about expensing one |
| 07:21.33 | xous | the plantronics headset |
| 07:21.45 | xous | heh. I ain't paying $200-300 out of pocket :P |
| 07:22.33 | drmessano | Why are you going to move, of all people, the RECEPTIONIST to a soft client? |
| 07:22.33 | *** join/#asterisk JamesO (~JamesO@195.66.232.93) |
| 07:22.44 | drmessano | Wouldnt you want her phone to work even if her PC craps out? |
| 07:22.49 | drmessano | Like, her, of all people? |
| 07:23.16 | xous | uhh. well she already is tied to her computer anyway |
| 07:23.27 | xous | we answer calls for multiple businesses |
| 07:23.27 | drmessano | How so? |
| 07:23.34 | drmessano | ok? |
| 07:23.45 | xous | so the only way she knows how to answer the phone is the prompt that shows up on the software on her computer |
| 07:24.10 | drmessano | Youre not using Asterisk? |
| 07:24.20 | xous | one call can be "thanks for calling Intel, how may we help you?" |
| 07:24.53 | xous | the next can be "You've reached Apple...." |
| 07:25.03 | xous | not for that no |
| 07:25.03 | drmessano | You know you can prefix her CallerID to show which company is getting the call.. |
| 07:25.07 | [TK]D-Fender | Yes and you can get phones with wonderful displays on the. Displays you can mod the cALLERID ON so she knows how to answer |
| 07:25.14 | [TK]D-Fender | Or use separate registrations. |
| 07:25.20 | drmessano | That too |
| 07:25.26 | [TK]D-Fender | Or an on-screen display pushed to the phone's interface. |
| 07:25.34 | [TK]D-Fender | Or an autdio prompt upon answering. |
| 07:25.38 | xous | I might be able to show "Apple" or "Intel" |
| 07:25.44 | drmessano | A |
| 07:25.45 | drmessano | I |
| 07:25.46 | [TK]D-Fender | or a dozen other things I could come up with |
| 07:25.53 | drmessano | or the audio prompt |
| 07:26.00 | drmessano | or a distinctive ring |
| 07:26.02 | xous | yeah, but the WAY she handles the call is completely different |
| 07:26.07 | [TK]D-Fender | Take a MINUTE and see how many more you can come up with. |
| 07:26.15 | xous | for example for one company she takes food orders |
| 07:26.22 | [TK]D-Fender | <xous> yeah, but the WAY she handles the call is completely different <- just needs to know how to answer. |
| 07:26.22 | drmessano | Okay? |
| 07:26.38 | [TK]D-Fender | that is easy to identify multiple ways on a varienty of phones |
| 07:26.48 | [TK]D-Fender | and that's OUTSIDE of super-easy CID manipulation |
| 07:26.51 | xous | there are like pages of different responses based on what theyw ant |
| 07:26.52 | [TK]D-Fender | But.. whatever |
| 07:26.56 | [TK]D-Fender | bed time here.... |
| 07:26.58 | drmessano | LOL |
| 07:27.02 | [TK]D-Fender | ZZZZZZZZzzzzzzzzzzzzzzz |
| 07:27.10 | xous | and ~1000 different companies |
| 07:27.24 | drmessano | This sounds like that guy from the philipines that had that shady as shut business operation |
| 07:27.42 | drmessano | Your receptionist takes calls for ~1000 different companies? |
| 07:27.42 | xous | heh it's just a virtual office business |
| 07:27.50 | xous | well 6 of them do |
| 07:27.57 | xous | yes |
| 07:28.17 | drmessano | Are you in the Philippines? |
| 07:28.29 | xous | nope, even worse... Canada. |
| 07:28.48 | drmessano | Were you in the philippines years ago? |
| 07:28.52 | xous | nope |
| 07:29.07 | drmessano | O.o |
| 07:29.09 | xous | there are several very large business that do the same stuff. Regus? |
| 07:29.36 | drmessano | No, some shady as hell guy that was doing shady as hell stuff, and he even smelled shady over IRC |
| 07:29.51 | xous | heh. like what? |
| 07:29.56 | drmessano | The kind of guy that would like and say he was in Canada |
| 07:30.02 | drmessano | lie |
| 07:30.26 | drmessano | You dont mind if he let the dog sniff your IRC client, do you? |
| 07:30.37 | drmessano | Just need to check things out first |
| 07:30.40 | xous | lol you won't find anything interesting |
| 07:30.48 | xous | running irssi from AWS |
| 07:30.55 | drmessano | lol |
| 07:31.01 | drmessano | Youre not him then |
| 07:31.19 | xous | nope. |
| 07:31.20 | drmessano | He was using mIRC or MS Chat or something douchey |
| 07:31.30 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 07:32.33 | xous | heh. $oldjob had a client in California |
| 07:32.59 | xous | was doing one of those forclosure scams |
| 07:33.34 | xous | where they'd take 2k from people that were in forclosure and then not try and get them a new lender |
| 07:33.58 | xous | they'd order 100 new DIDs/mo then kill the old ones |
| 07:34.00 | drmessano | WHOA |
| 07:34.05 | drmessano | thats sounds like the guy |
| 07:34.21 | xous | they were so fucking dirty |
| 07:34.28 | xous | owner was a friend of the CEO |
| 07:34.52 | xous | so all I could do was nickle and dime them for everything when shit broke |
| 07:34.56 | xous | was rather amusing |
| 07:35.44 | drmessano | OH OH OH.. He was running some company that sold some package for people to buy foreclosure homes. What they sold them was actual SHIT.. and the phone bank was idiots who couldn't really help the useless product |
| 07:36.07 | xous | ah yeah, that shit is pretty common |
| 07:36.16 | xous | selling old lists of shit |
| 07:36.25 | drmessano | THATS what it was.. he had remote agents who worked from home |
| 07:36.32 | drmessano | Using softphones |
| 07:36.44 | drmessano | God I wanted to punch that guy remotely |
| 07:37.12 | drmessano | He'd pop in like every now and then like a bad sitcom character |
| 07:37.30 | xous | heh. well to be fair the people buying this shit were looking to profit from someone losing their home |
| 07:37.33 | drmessano | Had some new agent with NAT issues.. wanted to brag about the money he was making in the tax haven of the philippines |
| 07:37.54 | xous | not a lot of sympathy for someone like that doesn't do due diligence |
| 07:37.56 | drmessano | No, I offer no fairness |
| 07:38.16 | xous | if not fair then... at least karma?/ |
| 07:39.27 | xous | the more I do business with people the more I realize that there a lot of people that will do anything to fuck you out of a buck. |
| 07:42.14 | drmessano | Karma, maybe |
| 07:42.44 | drmessano | I have much lower requirements for karma |
| 07:44.17 | xous | hehe. I have a buddy who outted two business partners that were stealing from the two I worked for |
| 07:44.39 | xous | they had two sides to the business. HPBX and Parts |
| 07:44.50 | xous | Parts would sell all kinds of shit, old nortel shit, etc |
| 07:45.49 | xous | buddy was looking to buy the business and noticed that purchases for nortel shit were listed as expenses for the HPBX side of the business |
| 07:45.56 | xous | and the income was recorded under the parts |
| 07:46.45 | xous | HPBX did all Cisco phones. No reason whatever for them to buy nortel heh |
| 07:53.26 | drmessano | That sounds about right. I've heard of lots of that.. on the news |
| 07:58.12 | xous | heh |
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| 12:29.26 | *** join/#asterisk [sID] (~sid@77.81.226.38) |
| 12:29.50 | [sID] | Hi |
| 12:30.38 | [sID] | which may be the reason that I have the answer status and billsec shows 0 or - seconds |
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| 12:36.44 | Samot | 0 means unanswered. |
| 12:36.55 | Samot | I've never seen a negative value. |
| 12:42.22 | WIMPy | Unless you round up it can also mean that the call lasted less than a second. |
| 12:43.15 | *** join/#asterisk Martin` (~martin@shell.ipv6.octocore.net) |
| 12:56.05 | [sID] | I have a lot of records where I have given -40, -60, -118 |
| 12:56.50 | *** join/#asterisk defswork (~aporter@212.38.66.116) |
| 12:56.52 | [sID] | Samot: Unanswered? This should not be a NO ANSWER? |
| 12:57.59 | WIMPy | What records are we talking about? |
| 13:01.29 | [sID] | Of those that have billsec 0 and status ANSWERED |
| 13:02.42 | Samot | Well perhaps those are voicemails, they got the greeting and hung up. |
| 13:02.52 | Samot | I'm not a 100% on Asterisk CDRs rounding factors. |
| 13:03.08 | Samot | But again, I've never seen negative values. |
| 13:03.26 | Samot | What is the duration value you showing? |
| 13:03.35 | Samot | -you |
| 13:05.05 | [sID] | Samot: duration 67 billsec -40 |
| 13:05.27 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
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| 13:08.33 | WIMPy | Did anyone even say cdr? |
| 13:10.15 | [TK]D-Fender | WIMPy, "which may be the reason that I have the answer status and billsec shows 0 or - seconds" <- these are CDR fields being discussed |
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| 13:16.26 | [sID] | in a moment I will give more details |
| 13:17.07 | [sID] | etc duration -3 billsec -18 |
| 13:17.28 | [sID] | duration 15 billsec -16 |
| 13:18.03 | [sID] | a distosition = answered |
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| 13:19.36 | Samot | What version of Asterisk? |
| 13:22.30 | [sID] | Asterisk 11.7.0 |
| 13:22.45 | Samot | Are the start values correct? |
| 13:23.07 | Samot | Duration and billsec are determined by start/answer/end time values. |
| 13:23.36 | [sID] | ok, wait checked |
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| 13:30.46 | Samot | ? |
| 13:31.18 | [sID] | I have a lot of records and retrieves data |
| 13:32.49 | Samot | But you just need to look at the records you're already giving us data from. |
| 13:33.09 | Samot | Is the timestamp in the start field correct? |
| 13:33.42 | Samot | The only way you're going to get negative values is if the END time is less than the START/ANSWER times. |
| 13:34.28 | Samot | END - START = duration, END - ANSWER = billsec |
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| 13:39.23 | *** join/#asterisk blinky_ (~damia@host81-136-203-112.in-addr.btopenworld.com) |
| 13:41.19 | blinky_ | Hi all, I was wondering if someone could tell me how I would find out if the trunk between the asterisk box and the provider or the trunk between two remote sites is secure. I have inherited controll over two sites and am thinking there is no encryption running. Is it set as standard or all all the voice calls and passwords being transmitted unsecured? |
| 13:42.51 | [TK]D-Fender | "am thinking there is no encryption running" <- do you SEE encryption settings in your peers? Do you see it negotiated in your calls? |
| 13:44.12 | blinky_ | I don't see anything that is way I am asking. Are the calls being transmitted across the network so that someone with wireshark could catch them and listen to the call content? |
| 13:44.24 | [TK]D-Fender | If they aren't there then it certainly isn't more secure than a standard call. |
| 13:44.53 | blinky_ | That's what I thought. |
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| 14:02.32 | Samot | In terms of passwords. If you are referring to the SIP user/password being transmitted plain text during auth, no. |
| 14:03.14 | Samot | Now if you're using a provisioning server... |
| 14:03.40 | Samot | That can transmit that information insecurely but that can be secured as well. |
| 14:07.30 | [sID] | Samot: Ok, |
| 14:07.31 | [sID] | Yes, this is just how it comes out? |
| 14:07.33 | [sID] | Suspension asterisk? |
| 14:07.53 | [sID] | 2016-11-09 11:06:03 (START) |
| 14:07.53 | [sID] | 2016-11-09 11:06:32 (TALKING) |
| 14:07.53 | [sID] | 2016-11-09 11:06:23 (STOP) |
| 14:08.47 | Samot | Why is TALKING less than STOP? |
| 14:08.57 | Samot | Swap that |
| 14:09.03 | Samot | STOP < TALKING |
| 14:09.06 | blinky_ | At the moment I have 3 ip addresses trying to register sip extensions and failing with the wrong passwords. The issue is that 2 are from paris and the third is from the US. Our servers do not have users in any of those locations. |
| 14:09.21 | Samot | Then block those IPs. |
| 14:09.42 | blinky_ | We are in the process of that, at the same time we would like to know how they got in |
| 14:09.54 | WIMPy | Yes, they are the usual locations. |
| 14:10.06 | WIMPy | But luckily the most SIP attacks are easily filtered. |
| 14:11.00 | blinky_ | This is a little more serious as we do not have external users. The servers are used onjly by users on the local network. |
| 14:11.35 | blinky_ | So they came in either through the trunk or some other way, this is why we must find the hole fast |
| 14:13.08 | Samot | That's not how a trunk works. |
| 14:13.25 | Samot | A trunk says "This is allowed" |
| 14:13.52 | [sID] | Samot: |
| 14:13.53 | [sID] | This has just do not know. |
| 14:13.54 | [sID] | I think that may be the reason |
| 14:14.13 | WIMPy | Looks like you need a serious conversation with your firewall (admin). |
| 14:14.18 | Samot | You'll have to explain it again. |
| 14:14.35 | Samot | Because I didn't follow what you just said as the reason. |
| 14:16.24 | blinky_ | He left, I am trying to close all the holes now. This has only been found because people complained about call quality. When I looked into the logs I found the IP addresses. I had a crash course in asterisk 1 day ago, google is great for crash courses. |
| 14:17.33 | [TK]D-Fender | Samot> Now if you're using a provisioning server... <- doesn't change the SIP security, just means that something else could be making things WORSE by sending configs over the internet. Except tthat provisioning COULD using a secure communication metthod (or not). So not really much to speak for :) |
| 14:18.57 | [TK]D-Fender | blinky_> We are in the process of that, at the same time we would like to know how they got in <- they'd get in for not having been blocked. They probably didn't randomly decode a password rather than successfully brute-force (assuming they used a peer that auths) |
| 14:19.07 | Samot | I agree |
| 14:19.24 | [TK]D-Fender | Shich means they had enough shots at it that you should have been able to catch them to block the subsequent one that got them in |
| 14:21.10 | Samot | You have allowguests=no ? |
| 14:21.36 | [sID] | Samot: and how is the answer and billsec 0 what means? |
| 14:22.13 | [sID] | disposition answered billsec 0 |
| 14:22.38 | [TK]D-Fender | if you answer and hangup within a second that could make sense |
| 14:24.23 | [sID] | ok, thx for help |
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| 15:04.05 | tompaw | Morning |
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| 15:04.37 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 15:05.21 | tompaw | So, I managed to confirm with almost-absolute certainty that it's the act of *answering* the 1st channel in a bridge prior to processing Dial() on the 2nd channel that overwrites the 2nd channel's CLI. |
| 15:05.46 | igcewieling | egads, more Asterisk boxes hacked or attacked? |
| 15:06.01 | tompaw | The question now is: how to get around it without delaying channel 1's Answer. |
| 15:07.11 | WIMPy | Yes. Answering sets CONNECTEDLINE. |
| 15:09.14 | tompaw | WIMPy: I've added CONNECTEDLINE to my dialplan: https://hastebin.com/nuyivixuja.rb |
| 15:10.11 | tompaw | and yet still this Dial() from line 9 uses wrong CLI |
| 15:10.49 | WIMPy | Does it happen before or after the 1st channel is answered? |
| 15:11.39 | tompaw | Here's the catch - I don't know. These are two async processes. Look at line 14 and 21 - these are called one after another, but are both async calls. |
| 15:11.51 | WIMPy | You could try to set it yourself via a gosub on answer. |
| 15:12.46 | tompaw | You mean in channel 1? |
| 15:13.06 | WIMPy | yes |
| 15:13.18 | tompaw | Hm... I have not tried that. |
| 15:13.27 | WIMPy | Well, or both, unless you know which one is answered first. |
| 15:14.01 | WIMPy | Or you could try confbridge instead of a simple bridge that shouldn't forward any connected line information. |
| 15:14.27 | tompaw | That's going to be a 50/50, because as you can see the 2nd one's dialplan begins with Answer(), but at the same time the first one hits an angularjs CRM that answers it immediately. |
| 15:14.35 | WIMPy | Should. IIRC it once did. |
| 15:14.57 | tompaw | Erm... can ari do confbridges? |
| 15:15.35 | WIMPy | I have no idea. But I would assume so. |
| 15:17.21 | tompaw | I'm gonna simplify this pastebin a little bit and ask the folks at #ari |
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| 15:20.05 | *** join/#asterisk Stese (52467951@gateway/web/cgi-irc/kiwiirc.com/ip.82.70.121.81) |
| 15:22.19 | Stese | Hi All, |
| 15:23.28 | *** join/#asterisk Rini (uid196547@gateway/web/irccloud.com/x-elcrhpwfeangujst) |
| 15:23.37 | Stese | I'm having issues with no inbound audio on a new freepbx install... Obviously i'm looking at NAT being the cause, but I'm not seeing any errors in the asterisk CLI... is this common? |
| 15:24.44 | [TK]D-Fender | That you're not seeing anything? |
| 15:24.53 | [TK]D-Fender | We don't know what you're seeing. |
| 15:25.11 | [TK]D-Fender | So we can't tell you if you're looking properly or simply missing something tthatt might be tthere |
| 15:25.22 | *** join/#asterisk scgm11_ (~nobody@ec2-52-207-226-93.compute-1.amazonaws.com) |
| 15:25.32 | Stese | is it common not to get errors on Asterisk CLI, when dealing with a NAT issue... |
| 15:25.48 | [TK]D-Fender | I can tell you that TTHIS is common (people not looking, or interpreting properly even when they do). |
| 15:25.49 | igcewieling | Stese: correct. no audio is not an error. |
| 15:26.31 | igcewieling | have you set the correct settings for localnet=, externip=, and nat= settings? |
| 15:26.33 | Stese | OK, thanks. |
| 15:26.42 | Stese | igcewieling: Yes, |
| 15:26.51 | [TK]D-Fender | Show us the failed call |
| 15:26.58 | [TK]D-Fender | ? pb |
| 15:27.02 | [TK]D-Fender | ~pb |
| 15:27.08 | infobot | [pastebin] 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://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 15:27.11 | [TK]D-Fender | ^^^ |
| 15:27.12 | Stese | I know how topastebin :P |
| 15:27.18 | [TK]D-Fender | Making sure. |
| 15:27.22 | Stese | but not use spacebar correctly :P |
| 15:28.00 | [TK]D-Fender | Well you seem keen on correcting mistakes, so there is hope for you... |
| 15:29.17 | Stese | ;). I've come back to asterisk after a few years not working in a place that was even remotely interested... so it's all a bit different. |
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| 15:42.57 | Stese | Here's that 'failed' call, after setting the verbosity correctly.... |
| 15:43.00 | Stese | http://pastebin.com/CV1nFUhv |
| 16:01.39 | [TK]D-Fender | You aren't looking at SIP debug at all which qualifies as "not even looking properly" |
| 16:01.58 | [TK]D-Fender | So yup, we're in "scenario #1" |
| 16:02.14 | [TK]D-Fender | "pjsip set logger on" |
| 16:02.18 | [TK]D-Fender | Go try anotther call |
| 16:02.35 | Stese | Ahh, sorry, thought you just wanted the verbose output. 1 sec |
| 16:03.04 | *** join/#asterisk friedrich (~friedrich@aextron.de) |
| 16:03.34 | igcewieling | Stese: few people are willing to look through FreePBX CLI logs. |
| 16:04.01 | [TK]D-Fender | That doesn't qualify anything when trying to pin where you (or the other side) is telling the other where to send audio. |
| 16:04.40 | [TK]D-Fender | Step 1: prove your side i sending the right one. Step 2: ensure that if you're supposed tto trust them that you do, and if you don'tt that you aren't taking what they offer att face value |
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| 16:09.27 | Stese | http://pastebin.com/NmqbJGFR |
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| 16:15.31 | igcewieling | looks to me like NAT is not setup in Asterisk correctly. o=300 8000 8000 IN IP4 10.73.34.30 |
| 16:16.00 | igcewieling | IIRC that should be whatever externip= was set, though I'd have to research it to be sure. |
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| 16:18.28 | igcewieling | I see you have g279 allowed. dont' do that when troublehsooting. |
| 16:19.05 | Stese | igcewieling: just so I can find the bit you are looking at in the paste, did you note what line? |
| 16:19.28 | Stese | igcewieling: ok, i'll disable it. any reason why? |
| 16:20.06 | igcewieling | In case there are problems with your G729 licenses |
| 16:20.54 | igcewieling | Hard to tell, but I might have been looking at the invite from the phone to asterisk. |
| 16:21.18 | igcewieling | My bet is the wrong externip. |
| 16:21.33 | Stese | g729 isn't enabled as far as I can see. |
| 16:22.34 | igcewieling | I must have been looking at the phone invite, so it doesn't matter. |
| 16:24.09 | igcewieling | stop removing all the important info like IPs. |
| 16:25.10 | Samot | Why are you offering up g711u first when your provider only seems to be offering g711a |
| 16:26.32 | Stese | Samot: I've no idea, this is an out of box FreePBX install. |
| 16:26.50 | Samot | Well, one you're in the wrong channel. |
| 16:26.50 | Stese | the only 2 codecs we have enabled, are ulaw and alaw |
| 16:27.02 | Samot | I know. |
| 16:27.14 | Samot | You're offering up both on the trunk, the provider only offers one.. |
| 16:27.42 | Samot | If the provider doesn't offer/support ulaw, having it as the primary codec in the trunk peer is really not needed. |
| 16:27.48 | [TK]D-Fender | <igcewieling> IIRC that should be whatever externip= was set, though I'd have to research it to be sure. <- this is PJSIP |
| 16:27.54 | [TK]D-Fender | nott chan_sip |
| 16:28.07 | [TK]D-Fender | Codecs don't lead to "no audio" |
| 16:28.14 | igcewieling | [TK]D-Fender: ah, they use totally different terminalogy for that item? |
| 16:28.27 | [TK]D-Fender | Codec mismatch leads to "call instantly dropped like a rock with extreme prejudice" |
| 16:28.35 | Samot | ^ That yes. |
| 16:28.40 | igcewieling | what IS the term? |
| 16:28.41 | [TK]D-Fender | igcewieling, Yes, and things one can do the other can't |
| 16:28.47 | Samot | But having un-needed stuff is just pointless. |
| 16:28.55 | Stese | Just to be clear, the call connects and outbound audio is fine. |
| 16:29.27 | [TK]D-Fender | ;external_media_address=203.0.113.1 |
| 16:29.27 | [TK]D-Fender | ;external_signaling_address=203.0.113.1 |
| 16:29.34 | Stese | Samot : I'll clear it up later. I've landed in this job last week, and they want me to get this running from the previous guys attempts |
| 16:29.34 | [TK]D-Fender | as per the sample config |
| 16:29.41 | Samot | Yes, no inbound audio is generally a NAT issue. |
| 16:29.50 | igcewieling | [TK]D-Fender: he SAID he had it set. |
| 16:30.06 | [TK]D-Fender | Do we see configs proving that? |
| 16:30.14 | igcewieling | so now everything he said is suspect |
| 16:30.50 | [TK]D-Fender | SAID? HA! **PROVE** |
| 16:31.10 | Stese | I've set it through the interface... |
| 16:31.29 | Stese | but, tbh, i don't trust it all that much, gonna look at the config files. |
| 16:31.35 | igcewieling | Stese: if you want to use the GUI go it #FreePBX. |
| 16:32.02 | *** part/#asterisk igcewieling (~ewieling@ip98-170-211-145.pn.at.cox.net) |
| 16:32.24 | Stese | igcewieling: Thanks, I'll ask there if I get stuck further. I'm going to check the externip issue |
| 16:40.43 | [TK]D-Fender | I also haven't seen it in GUI screen-shots.... |
| 16:41.16 | Stese | I can get them, no worries... |
| 16:42.03 | Stese | but here isn't the place for them |
| 16:47.50 | Stese | http://pastebin.com/6EsNjpfA < external address shown |
| 16:50.10 | Stese | http://pastebin.com/gQbRcSAe < ExternIP shown. |
| 16:50.51 | Stese | weirdly, nat=force_rport,comedia rather than nat=yes |
| 16:51.04 | Stese | would that be the issue? |
| 16:54.37 | [TK]D-Fender | taht is chan_sip, not pjsip |
| 16:54.49 | [TK]D-Fender | those configs have NOTHING to do with your calls |
| 16:55.37 | Stese | Ok, i'll see if the pjsip config files contain anything similar... |
| 16:57.34 | [TK]D-Fender | ~assume |
| 16:57.34 | infobot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav It makes an (ass) out of (u) and (me) |
| 17:06.07 | *** join/#asterisk moreder (~moreder@88-222-225-208.meganet.lt) |
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| 19:51.05 | *** join/#asterisk Brian1001 (~Brian1001@145.133.16.52) |
| 19:51.07 | Brian1001 | hi everyone |
| 19:51.26 | Brian1001 | Is there an Expert here who managed to configure Asterisk to a broadworks platform |
| 19:51.42 | Brian1001 | my provider uses broadworks and geez its a pain , i keep getting 403 forbidden when i try to call outside |
| 19:52.22 | [TK]D-Fender | Show us the call |
| 19:53.24 | Brian1001 | [2017-01-09 00:05:11] WARNING[6635][C-00000003]: chan_sip.c:23862 handle_response_invite: Received response: "Forbidden" from '<sip:08504XXXXX@domain.voip.nl>;tag=as1d8daea8' |
| 19:53.55 | Brian1001 | i cant just use SIP username / password , sip server, my provider seems to use a proxy |
| 19:54.11 | Brian1001 | so it took me an hr to even get registered |
| 19:55.52 | [TK]D-Fender | the actual full debug. |
| 19:55.58 | [TK]D-Fender | that summary statementt won't cutt it |
| 20:24.15 | Brian1001 | i managed to get it working somehow using this workaround : http://asteriskfaqs.org/2012/08/18/asterisk-users/make-outgoing-calls-through-broadworksbroadsoft-sip-gateway-from-asterisk.html |
| 20:24.17 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-vqbxqebxialdxqmr) |
| 20:24.17 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 20:26.51 | Samot | What version of Asterisk? |
| 20:27.13 | WIMPy | Thats a commpn issue. Not only with Broadcom. |
| 20:27.21 | Samot | Because I used to see this with Asterisk and BroadWorks, but like 4-5 years ago. |
| 20:28.00 | Brian1001 | Asterisk 13.12.1 |
| 20:28.19 | Brian1001 | i could use my Fritz!box as a proxy, the fritz1box connects perfectly |
| 20:28.26 | Brian1001 | and let asterisk connect to my router (fritzbox |
| 20:28.44 | Samot | OK, current versions should have much issue. |
| 20:28.54 | Samot | I would see it on 1.4 and 1.6 most the times.. |
| 20:29.11 | Samot | It's the digest support.. |
| 20:29.34 | WIMPy | Or just a missing proxy. |
| 20:29.41 | Brian1001 | yes the proxy i think |
| 20:29.50 | Samot | Well if they are using a proxy, sure. |
| 20:30.08 | *** join/#asterisk miralin (~Thunderbi@194.8.128.48) |
| 20:30.10 | Samot | I'm saying I saw this with Asterisk connecting to BW when I was using BW at the Teleco. |
| 20:30.13 | Brian1001 | "Forbidden" from '<sip:user02@MY_IP_ADDRESS>;tag=as21fcc482' |
| 20:30.19 | Samot | It was a digest issue. |
| 20:30.38 | Brian1001 | ah ok ! |
| 20:30.52 | Brian1001 | can you please give me a tiny example of what you mean |
| 20:30.54 | Samot | Brian1001: Don't get excited. |
| 20:30.56 | WIMPy | You still configure the host with the domain, but the proxy by IP. And don't forget a ",force" otherwise it won't helkp much. |
| 20:31.03 | Samot | The answer was to upgrade to 1.8 back then. |
| 20:31.17 | Brian1001 | a force |
| 20:31.34 | Brian1001 | i wonder where i can set the force |
| 20:32.30 | Brian1001 | my other provider uses portaOne and omg i was done within 5 min |
| 20:38.33 | Brian1001 | ok i think i know where it goes wrong , asterisk keeps getting : Forbidden" from '<sip:user01@MY_IP_ADDRESS but it should say user01@domainprovider |
| 20:39.42 | *** join/#asterisk tzafrir (~tzafrir@bzq-82-81-175-197.red.bezeqint.net) |
| 20:46.25 | Samot | fromdomain=domainprovider |
| 20:46.56 | WIMPy | that doesn't change the to:. |
| 20:47.17 | WIMPy | You have to put it in host=. |
| 20:47.41 | Brian1001 | yes i did host=domainprovider |
| 20:48.40 | Samot | Try fromdomain=domainprovider |
| 20:48.52 | Brian1001 | ok |
| 20:50.02 | Brian1001 | <PROTECTED> |
| 20:50.03 | Brian1001 | oh |
| 20:50.13 | Brian1001 | for the first time i now hear "the number you have dialed is not in service |
| 20:50.21 | Brian1001 | lol im happy that i hear something at least this time |
| 20:50.29 | Samot | That's a generic message on ASterisk. |
| 20:50.37 | Samot | We would need to see this call to tell you what actually happened. |
| 20:51.29 | WIMPy | Huh? I don't think that message esists. |
| 20:51.41 | Brian1001 | debug 255 ? |
| 20:51.45 | Samot | No. |
| 20:51.50 | Samot | Just do: asterisk -rvvvvvvvvv |
| 20:51.54 | Samot | Make the call. |
| 20:52.17 | Samot | ~pb |
| 20:52.17 | infobot | pastebin is, like, 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://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 20:52.23 | Samot | ^ put results there. |
| 20:52.28 | Brian1001 | ok |
| 20:52.31 | Brian1001 | will do that :) |
| 20:53.42 | Brian1001 | http://pastebin.com/Kh4YuxRA |
| 20:53.46 | [TK]D-Fender | "sip set debug on" <------------- |
| 20:54.00 | [TK]D-Fender | "core set verbose 0" |
| 20:54.12 | [TK]D-Fender | "core set verbose 10" |
| 20:54.16 | [TK]D-Fender | fix these and show a new call |
| 20:54.37 | [TK]D-Fender | Though I asked to see the call an HOUR ago... |
| 20:55.28 | Samot | Boom -> "1000@ext-local" |
| 20:55.45 | Samot | Totally FreePBX |
| 20:56.37 | Brian1001 | yes one sec, im using copy / paste |
| 20:56.47 | [TK]D-Fender | I don't care about it being freepbx here |
| 20:56.52 | Brian1001 | yes i've put it on a raspberry pi as a test , but .... |
| 20:56.52 | [TK]D-Fender | we'r talking sip peer auth |
| 20:57.00 | [TK]D-Fender | and those setttings are a dumb textt block |
| 20:57.06 | [TK]D-Fender | Let's just see it |
| 20:57.08 | Brian1001 | almost done with the new copy / paste |
| 20:57.32 | Brian1001 | http://pastebin.com/Su8uf5Wm |
| 20:57.56 | Brian1001 | SIP/2.0 484 Address incomplete |
| 20:59.03 | [TK]D-Fender | You left CORE on |
| 20:59.12 | [TK]D-Fender | massive garbe |
| 20:59.14 | [TK]D-Fender | look up |
| 20:59.17 | [TK]D-Fender | set those accordingly |
| 20:59.29 | [TK]D-Fender | garbage* |
| 20:59.33 | Samot | Yeah: core set debug 0 |
| 20:59.35 | Samot | Please. |
| 20:59.37 | Brian1001 | ok one sec |
| 20:59.37 | Brian1001 | yes |
| 20:59.45 | Brian1001 | sorry for that mistake |
| 20:59.59 | Brian1001 | Core debug was 255 and is now OFF. |
| 21:00.02 | Brian1001 | gonna call again xD |
| 21:00.21 | Samot | sip set debug on <-- do that |
| 21:00.44 | Samot | core set verbose 10 |
| 21:00.49 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 21:00.54 | Samot | Just to make sure. |
| 21:02.14 | Brian1001 | almost done |
| 21:02.33 | Brian1001 | new copy / paste : http://pastebin.com/XDxSsr7q |
| 21:03.07 | *** join/#asterisk tristero (~al.f.zero@unaffiliated/transfinite) |
| 21:04.39 | Brian1001 | SIP/2.0 484 Address incomplete |
| 21:04.40 | Brian1001 | hmmm |
| 21:06.07 | Brian1001 | SIP/2.0 401 Unauthorized |
| 21:06.45 | Samot | 088043674808000403 <-- What is this? |
| 21:07.10 | Brian1001 | oh that is wrong |
| 21:07.16 | Samot | That's what you are sending to them. Is that a valid format of how they want you to send calls? |
| 21:07.20 | Brian1001 | 088etc is my number UNTIL the 0800 |
| 21:07.27 | Brian1001 | 08000403 is a number i've tried to call |
| 21:07.42 | Brian1001 | no thats wrong , i seriously only typed in : 08000403 |
| 21:07.50 | Brian1001 | the 1st part is my own no |
| 21:08.09 | Samot | INVITE sip:08000403@192.168.178.43 SIP/2.0 <- You sent this to the PBX |
| 21:08.27 | Brian1001 | ok |
| 21:08.31 | Brian1001 | so the prolbem is the PBX |
| 21:08.38 | Brian1001 | because xLite does sent teh right number |
| 21:08.40 | Samot | INVITE sip:088043674808000403@iphonedok.tfzvoip.nl SIP/2.0 <-- This gets sent to the provider. |
| 21:08.48 | Samot | No. |
| 21:08.53 | Samot | X-Lite is sending the right number. |
| 21:08.57 | Samot | INVITE sip:08000403@192.168.178.43 SIP/2.0 <- You sent this to the PBX |
| 21:09.05 | Samot | ^ That's the number you dialed. |
| 21:09.08 | Brian1001 | yes |
| 21:09.09 | Samot | INVITE sip:088043674808000403@iphonedok.tfzvoip.nl SIP/2.0 <-- This gets sent to the provider. |
| 21:09.17 | Samot | ^ That's what the PBX sends to the provider. |
| 21:09.35 | Brian1001 | you're super sharp ! |
| 21:09.47 | Samot | Why are you prepending digits to it? |
| 21:10.22 | Brian1001 | i dont know |
| 21:10.29 | Samot | 0880436748 <-- This is being added before you dial the provider.. |
| 21:10.34 | Samot | What's in your Outbound Routes? |
| 21:10.38 | Brian1001 | im gonna check it |
| 21:13.12 | Samot | Reliably Transmitting (NAT) to 89.146.63.243:5060: |
| 21:13.22 | Brian1001 | this is OK |
| 21:13.25 | Samot | ^ Providers are generally not behind NAT |
| 21:13.33 | Brian1001 | the dial patern only has X. now |
| 21:13.34 | Brian1001 | testing again |
| 21:13.37 | Samot | No. |
| 21:13.51 | Samot | You were matching just fine. |
| 21:13.53 | Brian1001 | ok |
| 21:14.00 | Samot | If it didn't match it would send the call out. |
| 21:14.04 | Samot | You are PREPENDING |
| 21:14.07 | Brian1001 | ah yes true! |
| 21:14.10 | Samot | I.e. ADDING |
| 21:14.15 | Brian1001 | i wish i knew where |
| 21:14.22 | Samot | Look in the prepend field. |
| 21:14.44 | Brian1001 | ok one sec please (looking it up) |
| 21:15.08 | Samot | The first box on the left |
| 21:15.21 | Samot | Prepend | Prefix | Pattern | Caller ID |
| 21:15.37 | Brian1001 | OMG! |
| 21:15.41 | Brian1001 | that was the fix! |
| 21:15.48 | Brian1001 | Outbound Dial Prefix ..... |
| 21:15.56 | Brian1001 | that box is empty now and i can Call out !!! |
| 21:16.05 | Brian1001 | Samot you rock Mr! |
| 21:16.14 | Brian1001 | thank you so much |
| 21:16.23 | Samot | Well I think you required a "fromdomain" as well. |
| 21:16.37 | Brian1001 | yes ! |
| 21:16.39 | *** join/#asterisk slack1 (~slack@88-111-150-65.dynamic.dsl.as9105.com) |
| 21:16.53 | Brian1001 | i was getting that error right after using the fromdomain |
| 21:17.13 | Samot | 'fromdomain' was part of their validation scheme. |
| 21:17.35 | Samot | It's how I handle access via my proxy as well. |
| 21:17.57 | Brian1001 | ah |
| 21:18.11 | Samot | Not only does the TO domain have to be valid, the FROM domain has to be as well.. |
| 21:18.20 | Brian1001 | its more secure ? |
| 21:18.22 | Samot | If it's not set, not allowed. |
| 21:18.31 | Samot | In a sense. |
| 21:19.09 | *** join/#asterisk Y04NN (~y04nn@2a01:e34:ef37:5870:bcaf:2cde:4e56:1c8a) |
| 21:19.17 | Samot | I don't care where it's from, if the FROM domain doesn't match one of my SIP domains you shouldn't be allowed through. |
| 21:19.32 | Brian1001 | oh cool! |
| 21:19.46 | Brian1001 | my other provider (the one who uses portaOne) doesnt even need all this |
| 21:19.57 | Samot | Generally those trying to exploit via SIP aren't setting from domains. |
| 21:20.06 | Samot | Some don't |
| 21:20.12 | Samot | It's all on how they do it. |
| 21:20.30 | Brian1001 | i almost gave up |
| 21:20.32 | Samot | Some use a combo of fromuser and fromdomain.. |
| 21:21.03 | Samot | So even if your SIP user is "user02" |
| 21:21.16 | Samot | And you're given "1238481" as the fromuser..... |
| 21:21.22 | Brian1001 | correct thats what this provider wants as well! |
| 21:21.28 | Samot | Right. |
| 21:23.11 | Brian1001 | just incoming and i have a working situation |
| 21:25.20 | *** join/#asterisk DoYouKnow (~DoYouKnow@unaffiliated/doyouknow) |
| 21:25.31 | DoYouKnow | does asterisk compile on arm with the arm enhancements for DSP? |
| 21:25.38 | DoYouKnow | or do I have to do somethin special for that? |
| 21:25.40 | cresl1n | No |
| 21:25.48 | cresl1n | well, it compiles on ARM |
| 21:25.58 | cresl1n | I don't *think* it has ARM specific ASM to make it run quicker |
| 21:26.49 | DoYouKnow | hmm ok |
| 21:26.58 | DoYouKnow | I hope it runs well on the orange pi pc |
| 21:27.06 | DoYouKnow | I only have 1-2 lines I'll be using it with though |
| 21:27.10 | DoYouKnow | and google voice |
| 21:27.22 | DoYouKnow | do you think it will work? |
| 21:27.29 | cresl1n | yes |
| 21:27.33 | cresl1n | probably |
| 21:27.35 | DoYouKnow | ok |
| 21:43.02 | DoYouKnow | I probably should set some kind of resource limits on everything except asterisk |
| 21:43.26 | DoYouKnow | is there a premade config for doing that once installed? |
| 21:43.43 | DoYouKnow | give asterisk like a realtime priority |
| 21:46.42 | DoYouKnow | just set the nice? |
| 21:53.18 | *** join/#asterisk tuxian (~tuxian@igilmour.plus.com) |
| 22:04.13 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:07.33 | *** join/#asterisk clopez (~tau@neutrino.es) |
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| 22:35.23 | mcargile | can the Asterisk Wait CMD wait 0.1 seconds or does it need to be an integer? |
| 22:37.01 | [TK]D-Fender | "core show application wait" |
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| 23:42.39 | *** join/#asterisk infobot (ibot@rikers.org) |
| 23:42.39 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.13.1 (2016/12/08), 11.25.1 (2016/12/08), Standard: 14.2.1 (2016/12/08); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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