| 00:03.11 | ModFather | [TK]D-Fender it seems some times it works some time not |
| 00:03.34 | [TK]D-Fender | "it works" is again a useless way to look at this |
| 00:03.44 | [TK]D-Fender | There is no "it" |
| 00:03.55 | [TK]D-Fender | that CALL looks the way that it does when it comes in |
| 00:04.06 | [TK]D-Fender | If it is not matching then it is clearly DIFFERENT from the other calls |
| 00:04.14 | [TK]D-Fender | And you should be LOOKING AT THE CALL. |
| 00:04.24 | [TK]D-Fender | You are talking about the effect and not looking at the actual call that CAUSES it |
| 00:04.29 | [TK]D-Fender | LOOK AT THE CALL |
| 00:04.34 | [TK]D-Fender | And you'll see why it doesn't match |
| 00:04.37 | ModFather | when i call the number 2 |
| 00:04.41 | ModFather | it shows this: |
| 00:04.45 | Docfxit | I'm trying to find my asterisk.ctl file. I used this to find it sudo find / -name \asterisk.ctl It didn't find it. Do I have the syntax wrong? |
| 00:04.45 | ModFather | [Jan 11 00:02:47] NOTICE[13729][C-00000009]: chan_sip.c:25660 handle_request_invite: Call from 'number2' (54.172.60.2:5060) to extension '2222' rejected because extension not found in context 'sales'. |
| 00:04.55 | ModFather | after showing this error 2-3 times |
| 00:04.57 | [TK]D-Fender | ModFather, that is not SIP DEBUG |
| 00:05.00 | ModFather | then it connects to '3333' |
| 00:05.14 | ModFather | its seems like trying to find the correct one |
| 00:05.22 | [TK]D-Fender | There is no "find" |
| 00:05.27 | [TK]D-Fender | There is nothing to search |
| 00:05.28 | [TK]D-Fender | '2222' rejected because extension not found in context 'sales'. |
| 00:05.46 | [TK]D-Fender | That call matched whatever peer it matched, and got sent to [sales] because of it |
| 00:05.50 | ModFather | 2222 (general) is number1 and 3333 (sales) is number 3333 |
| 00:05.53 | [TK]D-Fender | And it was looking for 2222 in there |
| 00:06.08 | [TK]D-Fender | No |
| 00:06.17 | [TK]D-Fender | Your understanding of Asterisk is very screwed up |
| 00:06.45 | [TK]D-Fender | NUMBERS don't determine the context they are sent to |
| 00:06.57 | [TK]D-Fender | The section definition does |
| 00:07.33 | ModFather | it was just an EXAMPLE |
| 00:07.39 | ModFather | to let you understand |
| 00:08.23 | [TK]D-Fender | It's a broken example that says nothing useful |
| 00:11.33 | ModFather | [TK]D-Fender http://pastebin.ca/3326787 |
| 00:13.07 | ModFather | when calling the 1800000000 i get [Jan 11 00:12:27] NOTICE[13729][C-0000001c]: chan_sip.c:25660 handle_request_invite: Call from 'mygeneral' (54.172.60.1:5060) to extension '6666' rejected because extension not found in context 'TwilioCalling'. |
| 00:16.49 | [TK]D-Fender | ...and? |
| 00:17.35 | Docfxit | I'm trying to locate where my outgoing messages are located. I ran this: sudo asterisk -rvvvvvvv |
| 00:17.35 | Docfxit | <PROTECTED> |
| 00:17.47 | ModFather | sometimes i hear the twilio ( saying that i need to configure my number ) and sometimes it connects to through my asterisk and parse the call to the correct queue |
| 00:17.47 | [TK]D-Fender | Call matched the peer that it did and was send to 'TwilioCalling'. There is no match for 6666 in there just like it says |
| 00:18.07 | ModFather | the 6666 is on twilio |
| 00:18.21 | ModFather | i have set a sip trunk on a number i have buy |
| 00:18.23 | [TK]D-Fender | Docfxit, What is this "outgoing message" you're talking about? |
| 00:18.57 | Docfxit | When I call in I have a few outgoing messages. |
| 00:19.03 | [TK]D-Fender | <Docfxit> I'm getting an error saying: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist? <- it's telling you that either you don't have rights to read the PID file, or run asterisk to get to the CLI, or that * itself isn't actually running at all |
| 00:19.03 | ModFather | number1 to sip:5555@myserverhost and number2 to sip:6666@myserverhost |
| 00:19.12 | [TK]D-Fender | Docfxit, You're calling IN |
| 00:19.16 | [TK]D-Fender | that is INCOMING |
| 00:19.26 | [TK]D-Fender | Nothing magically goes "out" of your server |
| 00:20.21 | Docfxit | [TK]D-Fender, Asterisk is running. It's playing the wrong incoming message. |
| 00:20.51 | *** join/#asterisk bline (~bline@66-112-20-225.stat.centurytel.net) |
| 00:20.54 | [TK]D-Fender | Go properly connect to * cli |
| 00:21.05 | [TK]D-Fender | Actually do a FULL su |
| 00:21.10 | Docfxit | [TK]D-Fender, I looked into that folder for asterisk.ctl. The folder isn't there. |
| 00:21.15 | [TK]D-Fender | and prove your environment is set up right |
| 00:22.04 | [TK]D-Fender | * looks in the folder you tell it to for your config files including asterisk.conf which defines this. |
| 00:22.24 | [TK]D-Fender | if you installed into non-default folders then you have to sepcify where it is supposed to be found |
| 00:22.48 | [TK]D-Fender | Tell us something useful about how you installed your system and show us that * is runnig and that asterisk.conf is in a sane location |
| 00:22.51 | Docfxit | Great. I'll look at asterisk.conf to see what I find. Thanks. |
| 00:24.48 | ModFather | [TK]D-Fender do you know what it can cause it? |
| 00:25.51 | [TK]D-Fender | ModFather, You are not listening |
| 00:25.52 | Docfxit | [TK]D-Fender, The only thing I see in asterisk.conf is: ;astctl = asterisk.ctl And that's not being used. |
| 00:26.09 | [TK]D-Fender | ModFather, I've told you a dozen times now |
| 00:26.30 | [TK]D-Fender | Docfxit, Where IS the PID file? |
| 00:26.35 | Docfxit | [TK]D-Fender, My asterisk.conf is in: /etc/asterisk/ |
| 00:26.43 | [TK]D-Fender | [TK]D-Fender> Call matched the peer that it did and was send to 'TwilioCalling'. There is no match for 6666 in there just like it says |
| 00:26.55 | [TK]D-Fender | <[TK]D-Fender> Docfxit, Where IS the PID file? |
| 00:27.11 | [TK]D-Fender | Docfxit, Where is proof that * is running? How about the CLI parameters passed to it? |
| 00:27.47 | Docfxit | How could I find the PID file? |
| 00:27.50 | ModFather | [TK]D-Fender ofcourse is not match for 6666, because i was calling the number that i have define as 5555 |
| 00:28.44 | [TK]D-Fender | ModFather, not that I can see. Because I see NOTHING. You are showing NOTHING useful. |
| 00:29.11 | [TK]D-Fender | Docfxit, You say * is running SHOW US |
| 00:29.13 | Docfxit | [TK]D-FendeTk: I know * is running because when I call in the incoming messages are played. |
| 00:29.16 | [TK]D-Fender | SHOW US |
| 00:29.18 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 00:29.26 | [TK]D-Fender | Where the Process>? |
| 00:29.33 | [TK]D-Fender | Where's the proof of where it got launhed from? |
| 00:31.16 | ModFather | [TK]D-Fender http://d.pr/1kl1b/5rE1d3Ns |
| 00:31.40 | ModFather | i have configure from twilio the number to origin to: sip:6666@pbx.my-asteriskserver |
| 00:31.54 | ModFather | so it hits to my server |
| 00:32.02 | [TK]D-Fender | Where do you get the idea the name "mysales" MEANS ANTHING? |
| 00:32.15 | [TK]D-Fender | * matches the STUPID HOST IP the call is sent from |
| 00:32.18 | ModFather | its the name of pstn |
| 00:32.30 | [TK]D-Fender | What makes you think they SEND from a specific server IP based on your stupid phone number? |
| 00:32.33 | [TK]D-Fender | NOT |
| 00:32.34 | [TK]D-Fender | NO |
| 00:32.39 | [TK]D-Fender | IP ADDRESS <-- |
| 00:32.46 | [TK]D-Fender | That is how your call is matched |
| 00:32.54 | [TK]D-Fender | YOU don't get to tell Twilio what IP to send FROM |
| 00:32.58 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 00:33.09 | [TK]D-Fender | That name up top is GARBAGE |
| 00:33.12 | [TK]D-Fender | it means NOTHING |
| 00:33.34 | ModFather | if you look on the god damn logs i sant you on pastebin |
| 00:33.41 | ModFather | you will see that the name before .pstn.twilio.com |
| 00:33.44 | [TK]D-Fender | What logs? |
| 00:33.45 | ModFather | is mysales |
| 00:33.54 | ModFather | sorry i mean the sip.conf |
| 00:34.05 | ModFather | i sent you the sip.conf on the pastebin |
| 00:34.16 | [TK]D-Fender | so?> |
| 00:34.17 | [TK]D-Fender | garbage |
| 00:34.20 | [TK]D-Fender | means nothing |
| 00:34.23 | [TK]D-Fender | that isn't a LOG |
| 00:34.25 | [TK]D-Fender | that is not a CALL |
| 00:34.34 | ModFather | dude you are so so so rude |
| 00:34.40 | ModFather | good that you are a knowledger |
| 00:34.44 | ModFather | but you are so fucking rude |
| 00:34.45 | [TK]D-Fender | WHERE IS THE CALL? |
| 00:34.51 | [TK]D-Fender | I asked you for proof |
| 00:35.05 | [TK]D-Fender | You can have 10000 lines of config... and that does not mean the CALL will look like you think it should |
| 00:35.39 | [TK]D-Fender | And you don't get to tell twilio what IP they will use to call you from |
| 00:35.40 | Docfxit | [TK]D-Fender: After doing sudo asterisk -vvvvvvvvvv I placed the results in http://pastebin.ca/3326829 |
| 00:35.46 | [TK]D-Fender | Do you have a problem with this idea? |
| 00:35.58 | [TK]D-Fender | Docfxit, I told you to do a FULL su. |
| 00:35.59 | [TK]D-Fender | DO IT |
| 00:36.01 | [TK]D-Fender | and show uys |
| 00:36.03 | [TK]D-Fender | us |
| 00:36.19 | Docfxit | What is a full su? |
| 00:36.31 | [TK]D-Fender | su - |
| 00:36.37 | [TK]D-Fender | not SUDO |
| 00:36.52 | [TK]D-Fender | But from that... you are connectiing from what we last saw there |
| 00:36.58 | [TK]D-Fender | I don't see it "crashing out" |
| 00:37.04 | [TK]D-Fender | I also don't see a command prompt |
| 00:37.23 | [TK]D-Fender | so I can't tell if there is something we should have been seeing that you haven't shown us |
| 00:37.57 | Docfxit | su asterisk -vvvvvvvvvv |
| 00:37.57 | Docfxit | su: invalid option -- v |
| 00:38.02 | [TK]D-Fender | Also, connecting via that method should not be showing us "loading" messages like that unless someone is applying changes... or * is restarting in a loop, etc |
| 00:38.13 | [TK]D-Fender | Docfxit, I don't see the "r" in there <- |
| 00:38.19 | [TK]D-Fender | r= connect to CLI |
| 00:38.38 | cmendes0101 | "su -" |
| 00:39.06 | cmendes0101 | "su ..." will try and switch users |
| 00:42.21 | Docfxit | su asterisk -vvvvvvvvvvr |
| 00:42.21 | Docfxit | su: invalid option -- v |
| 00:42.29 | [TK]D-Fender | su - |
| 00:42.32 | [TK]D-Fender | THAT |
| 00:42.33 | [TK]D-Fender | ALONE |
| 00:42.38 | [TK]D-Fender | No "asterisk" after it |
| 00:42.58 | [TK]D-Fender | No extra garbage. |
| 00:45.35 | Docfxit | su - |
| 00:45.35 | Docfxit | su: Authentication failure |
| 00:47.03 | [TK]D-Fender | Don;'t know your root pass? |
| 00:49.30 | ModFather | take his head... |
| 00:49.47 | Docfxit | I thought I did. I guess I don't. How can I find it? |
| 00:49.55 | Docfxit | Or replace it. |
| 00:50.04 | cmendes0101 | reinstall os |
| 00:50.11 | ModFather | Docfxit , reinstall or use Rescue CD |
| 00:50.44 | [TK]D-Fender | or use a kernel boot option to force single-user |
| 00:50.52 | [TK]D-Fender | Or use sudo su - |
| 00:50.57 | [TK]D-Fender | And SET IT |
| 00:51.23 | [TK]D-Fender | Maybe tell us something useful about your working environment so we aren't guessing why your system works the way it does |
| 00:52.02 | Docfxit | I'm into the root now. How do I set the password? |
| 00:52.24 | Docfxit | What would you like to know about the working environment. |
| 00:52.34 | Docfxit | I'm running Ubuntu |
| 00:52.37 | ModFather | typeL passwd |
| 00:52.42 | ModFather | type: passwd |
| 00:52.42 | [TK]D-Fender | That's WHY then |
| 00:52.47 | [TK]D-Fender | And don't bother |
| 00:52.53 | [TK]D-Fender | Learn how linux works |
| 00:52.57 | [TK]D-Fender | Learn how ubuntu works |
| 00:53.02 | [TK]D-Fender | You picked your os. Learn it |
| 00:53.18 | [TK]D-Fender | Ubuntu specifically trys to STOP you from doing stuff as root as much as possible |
| 00:53.24 | [TK]D-Fender | and you didn't even PICK a root password |
| 00:53.35 | [TK]D-Fender | under desktop at least |
| 00:53.38 | [TK]D-Fender | is that what you're using? |
| 00:54.23 | Docfxit | I successfully change the root password. |
| 00:55.21 | [TK]D-Fender | before even bothering I'd lok at the rest of the setup |
| 00:55.24 | [TK]D-Fender | How did you install *? |
| 00:55.28 | [TK]D-Fender | Show us the PROCESS running. |
| 00:55.39 | [TK]D-Fender | Show us how you are STARTING it as a daemon in the first place |
| 00:55.47 | Docfxit | I'm currently in root. How can I get back to the user so I can put in su - |
| 00:55.53 | [TK]D-Fender | "exit" |
| 00:56.40 | Docfxit | su - takes me back to root. |
| 00:57.03 | [TK]D-Fender | "exit" |
| 00:57.17 | Docfxit | * Starts up automatically when I boot. |
| 00:57.22 | [TK]D-Fender | HOW? |
| 00:57.29 | [TK]D-Fender | WHat magically installed a script for this? |
| 00:57.35 | [TK]D-Fender | how did you install your system? |
| 00:57.52 | Docfxit | I had someone else install * years ago. I have no idea. |
| 00:58.23 | Docfxit | Where could I find an autostart script? |
| 00:58.36 | [TK]D-Fender | I sugget you hire a consultant as you ar missing basic linux awareness and awareness of what you have installed at all |
| 00:59.20 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 00:59.50 | Docfxit | [TK]D-FendeTk: Yes I am. I'm just trying to replace the incoming messages. I didn't want to make a huge project out of this. |
| 01:03.30 | [TK]D-Fender | Can you actually connect to CLI? |
| 01:03.45 | [TK]D-Fender | How was Asterisk configured? |
| 01:04.12 | [TK]D-Fender | I saw a PARTIAL attempt to connect to CLI that made your * install look like it might be CRASHING in a loop |
| 01:04.13 | Docfxit | How would I connect to CLI? |
| 01:04.24 | [TK]D-Fender | and I got no followup answer to questions about it |
| 01:04.29 | [TK]D-Fender | asterisk -rvvvvvvvvvvvvvv |
| 01:04.51 | [TK]D-Fender | do you GET THERE? Does it STICK? Does it CRASH out? |
| 01:04.58 | [TK]D-Fender | Or are you then SITTING in CLI? |
| 01:05.11 | [TK]D-Fender | Because before you couldn't even conenct and you gave us partial prrof |
| 01:05.17 | [TK]D-Fender | Answer this now |
| 01:05.39 | Docfxit | Everything looks fine until it gets to Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist? |
| 01:06.05 | Kobaz | sitting on the dock the cli |
| 01:06.14 | Kobaz | watching the channels gooooooooo awaaaaaay |
| 01:06.21 | [TK]D-Fender | Docfxit, You showed PARTIAL success earlier |
| 01:06.24 | [TK]D-Fender | What CHANGED? |
| 01:06.30 | [TK]D-Fender | Did that CRASH out? |
| 01:06.38 | Docfxit | This is as far as It got. |
| 01:07.11 | Docfxit | That's why I started asking where is my asterisk.ctl |
| 01:07.29 | Kobaz | Docfxit: yeah if you can't find your startup scripts, you need to get some basic linux stuff under your belt |
| 01:07.59 | Kobaz | Docfxit: depends on your system, if you're using systemd or basic sysV. /etc/rc* /etc/init.d/* is a good place to look |
| 01:08.58 | Kobaz | Docfxit: you shouldn't just randomly paste one line results in here... paste your entire console into a pastebin (with things like passwords and things removed for safety) |
| 01:09.02 | [TK]D-Fender | <Docfxit> [TK]D-Fender: After doing sudo asterisk -vvvvvvvvvv I placed the results in http://pastebin.ca/3326829 |
| 01:09.14 | Kobaz | okay that's a good start |
| 01:09.14 | [TK]D-Fender | I see a PARTIAL SUCCESS THERE |
| 01:09.16 | Kobaz | oh |
| 01:09.27 | Kobaz | Docfxit: people can't help you if you don't send everything they are asking |
| 01:09.36 | [TK]D-Fender | what the happened between you GETTING that far and FAILING now? |
| 01:09.50 | Docfxit | I'm trying to send everything I know how to. |
| 01:09.55 | Kobaz | Unable to bind socket: Address already in use |
| 01:10.05 | Kobaz | you have another asterisk running most likely |
| 01:10.16 | Kobaz | Docfxit: where's the rest of it? |
| 01:10.18 | Docfxit | I don't see how. |
| 01:10.21 | [TK]D-Fender | Hire a consultant. You have no idea what you're doing. This is a stupid amount of pain. |
| 01:10.25 | Docfxit | That's it. |
| 01:10.32 | Kobaz | that's not everything |
| 01:10.57 | Docfxit | I will find the start up script and get back. |
| 01:10.58 | *** join/#asterisk Oatmeal (~Suzeanne@75-103-145-152.ccrtc.com) |
| 01:22.57 | *** join/#asterisk esaym153 (~esaym153@75-1-182-8.lightspeed.snantx.sbcglobal.net) |
| 01:23.17 | esaym153 | anyone have any clue what causes issues like these missing module symbols: https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=780392 |
| 01:24.07 | esaym153 | been fighting it all day. Either something is wrong with my machine, or the person that wrote the compile rules for the debian packages got it all messed up... |
| 01:24.22 | esaym153 | I get the same for both 1.11 and 1.13 |
| 01:24.37 | esaym153 | with the debian stable package and the debian unstable packages |
| 01:54.43 | *** join/#asterisk boris_t (~boris_t@212.119.243.183) |
| 02:21.39 | esaym153 | brb |
| 02:33.47 | *** join/#asterisk Synthase_ (uid63346@gateway/web/irccloud.com/x-qhbuuxostszqeiij) |
| 02:57.02 | *** join/#asterisk vader- (~Adium@pool-71-175-67-97.phlapa.fios.verizon.net) |
| 02:57.32 | *** join/#asterisk amonk (~amonk@unaffiliated/amonk) |
| 03:00.29 | ModFather | Does anyone why CDR stores wrong calldate on db? |
| 03:02.32 | *** join/#asterisk oldhack (~oldhack@cpe-66-68-88-174.austin.res.rr.com) |
| 03:09.26 | *** join/#asterisk evil_gordita (robert@ip70-188-41-127.rn.hr.cox.net) |
| 03:10.21 | WIMPy | ModFather: YOu configured it to use UTC? Or your system settings are not what you want them to be. |
| 03:10.44 | ModFather | WIMPy i want to use my system time, but i see it stroes on db 00000 |
| 03:10.50 | ModFather | instead of any date time |
| 03:11.14 | WIMPy | Wrong type? |
| 03:13.28 | ModFather | http://d.pr/1aP0D/3yVWC3HR |
| 03:13.37 | ModFather | 00000000 |
| 03:14.13 | [TK]D-Fender | I'd prove the field is even MAPPED |
| 03:14.22 | [TK]D-Fender | And not jsut getting zeroed on add |
| 03:14.35 | [TK]D-Fender | And prove against other CDR methods including CSV |
| 03:15.30 | ModFather | it does same on CSV |
| 03:15.45 | ModFather | all fields okey except date |
| 03:16.36 | [TK]D-Fender | Show all the CDR configs, CLI status dumps for it and a CALL with CORE DEBUG jacked up so we can see what it is claiming to write |
| 03:20.08 | ModFather | http://pastebin.ca/3327009 ( all other fields are just saved fine ) |
| 03:27.19 | ModFather | core set debug 10 and core set debug 10 ? |
| 03:27.23 | ModFather | and then a call? |
| 03:27.33 | ModFather | i have also cdr set debug on |
| 03:27.56 | [TK]D-Fender | There are other configs if that is a DATABASE dump we're seeing |
| 03:28.21 | [TK]D-Fender | http://d.pr/1aP0D/3yVWC3HR <- which it very clearly looks like |
| 03:29.08 | poolson | hey i used to dial *78 for voicemail i think |
| 03:29.30 | poolson | now it seems to set DND |
| 03:29.49 | [TK]D-Fender | poolson, your dialplan does whatever you told it to do. It does not magically change |
| 03:29.54 | ModFather | [TK]D-Fender i am using default values i have touch only cdr odbc so i can store it to the database, |
| 03:29.56 | poolson | well yeah |
| 03:30.04 | poolson | the deeper question would be where to reset that ;) |
| 03:30.22 | [TK]D-Fender | ModFather, I asked for configs. You are not showing the very specific config related to that pic you showed us. |
| 03:30.35 | [TK]D-Fender | poolson, There is nothing to reset. It only would have changed if you changed it |
| 03:30.36 | ModFather | which configs you want to see? |
| 03:30.42 | [TK]D-Fender | poolson, vi extensions.conf |
| 03:30.59 | [TK]D-Fender | http://d.pr/1aP0D/3yVWC3HR |
| 03:31.12 | [TK]D-Fender | <[TK]D-Fender> Show all the CDR configs, CLI status dumps for it and a CALL with CORE DEBUG jacked up so we can see what it is claiming to write |
| 03:31.14 | ModFather | this is a screenshot from my Database |
| 03:31.56 | poolson | pretend im using a gui dude |
| 03:31.58 | poolson | im new ! |
| 03:32.04 | poolson | lol but yeah makes sense been reading about it |
| 03:33.01 | [TK]D-Fender | Pretend? |
| 03:33.10 | [TK]D-Fender | There is nothing to pretend here |
| 03:33.46 | [TK]D-Fender | FreePBX is *NOT* supported here and those questions should be asked in their channel: #freepbx |
| 03:34.16 | [TK]D-Fender | Good that you joined as I was writing that.... |
| 03:34.57 | poolson | hey fendor you got a carrot lodged in your ass |
| 03:35.01 | poolson | lighten up dude ! |
| 03:35.06 | poolson | fender |
| 03:35.14 | [TK]D-Fender | :) |
| 03:35.37 | [TK]D-Fender | I keep my #asterisk side Asterisk, and #freepbx side as FreePBX |
| 03:35.42 | [TK]D-Fender | ~freepbx |
| 03:35.42 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 03:35.49 | [TK]D-Fender | As it has always been here |
| 03:37.37 | poolson | yah its all good |
| 03:37.43 | poolson | i SOLVED IT ! |
| 03:39.44 | [TK]D-Fender | \o/ |
| 03:49.04 | WIMPy | ModFather: My solution to these kind of issues is pretty easy: Don't use a database, it's only another thing element to cause trouble. |
| 03:49.21 | ModFather | WIMPy indeed bro |
| 03:49.35 | ModFather | just added to my cdr_adaptive_odbc.conf |
| 03:49.36 | ModFather | alias start => calldate |
| 03:49.51 | ModFather | i am gonna try , if it doesnt work, i will stay without cdr in database |
| 03:50.01 | [TK]D-Fender | Databases work fine |
| 03:50.15 | [TK]D-Fender | Thosands of users use it for this all the time for more than a decade |
| 03:50.28 | WIMPy | Tehy hopefully do, but not every config does. |
| 03:50.52 | ModFather | my database is good, i double checked my fields etc.. i just missing something in my configuration |
| 03:51.08 | ModFather | WIMPy do you know the command i need to use on asterisk to see the variables it does sent to use INSERT? |
| 03:51.16 | ModFather | https://repositorioivanpaes.googlecode.com/files/command_asterisk_ok.PNG |
| 03:51.18 | ModFather | like this |
| 03:51.50 | WIMPy | Probably some debug level. |
| 03:52.07 | ModFather | which is the one you suggest? |
| 03:52.09 | WIMPy | They can be set per module to keep things readable. |
| 03:52.22 | ModFather | it worked by the way |
| 03:52.34 | ModFather | i just added that: alias start => calldate |
| 04:27.57 | *** join/#asterisk juned (~juned@202.131.119.122) |
| 04:28.15 | *** join/#asterisk rrittgarn (~rrittgarn@75-150-221-196-Illinois.hfc.comcastbusiness.net) |
| 05:55.24 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-86-39.hsd1.or.comcast.net) |
| 06:02.27 | *** join/#asterisk xochilpili (~xochilpil@unaffiliated/xochilpili) |
| 06:02.30 | xochilpili | hi there |
| 06:02.33 | xochilpili | :D |
| 06:08.04 | xochilpili | SIP/2.0 415 Unsupported Media Type ?? Zoiper and Linio also with Grandstream |
| 06:10.26 | *** join/#asterisk Aamit (~Amit@115.254.72.179) |
| 06:16.20 | xochilpili | ok |
| 06:16.26 | xochilpili | canreinvite=yes |
| 06:31.09 | *** join/#asterisk elitas (~elitas@213.226.135.203) |
| 06:54.34 | *** join/#asterisk goddva (~goddva@77.40.154.242) |
| 07:00.00 | *** join/#asterisk bof22 (~Thunderbi@185.13.183.107) |
| 07:03.44 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 07:14.34 | *** join/#asterisk Demon_VoIP (~demon@109.60.135.186) |
| 07:24.47 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 07:35.19 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 07:39.18 | *** join/#asterisk EOIP (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 07:56.07 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 07:58.18 | *** join/#asterisk lumasepa (~gestoip@193.145.124.30) |
| 07:58.32 | *** part/#asterisk lumasepa (~gestoip@193.145.124.30) |
| 08:03.20 | *** join/#asterisk Zogot (~Adium@185.21.52.255) |
| 08:16.17 | *** join/#asterisk juned (~juned@202.131.119.122) |
| 08:22.43 | *** join/#asterisk lambda79 (c3654885@gateway/web/freenode/ip.195.101.72.133) |
| 08:26.19 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:27.50 | *** join/#asterisk pchero_work (~pchero@109.70.54.56) |
| 08:41.48 | *** join/#asterisk Tim_Toady (~fuzzy@snf-33276.vm.okeanos.grnet.gr) |
| 08:42.21 | *** join/#asterisk zapata (~zapata@2a02:b18:581:10:cd3b:6f2a:462e:72ae) |
| 08:45.52 | *** join/#asterisk robink (quasselcor@unaffilated/robink) |
| 09:02.10 | *** join/#asterisk Tim_Toady (~fuzzy@snf-33276.vm.okeanos.grnet.gr) |
| 09:13.18 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 09:27.26 | *** join/#asterisk Jesterboxboy (~Thunderbi@chello080109194026.3.graz.surfer.at) |
| 09:46.05 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 09:49.42 | *** join/#asterisk sekil (~sekil@nat-73.net011.net) |
| 10:08.42 | *** join/#asterisk Tenhi_ (~tenhi@static-ip-69-64-50-196.inaddr.ip-pool.com) |
| 10:10.03 | *** join/#asterisk Demon_VoIP (~demon@109.60.135.186) |
| 10:14.14 | *** join/#asterisk Jesterboxboy (~Thunderbi@chello080109194026.3.graz.surfer.at) |
| 10:22.09 | *** join/#asterisk guest9001 (~bounceman@185.32.9.250) |
| 10:23.28 | guest9001 | can I match both +459100 and 458100 in the dialplan? |
| 10:23.40 | guest9001 | In case I do not know if the number comes in as E164 standard or not? |
| 10:28.22 | *** join/#asterisk kritzikratzi (~kritzikra@cpe90-146-150-86.liwest.at) |
| 10:29.16 | guest9001 | or can I jump to another exten in the same context? |
| 10:29.58 | Tim_Toady | sure, _+XXXXXX will match +459100 and _XXXXXX wil match 459100 |
| 10:30.01 | *** join/#asterisk defsdoor (~andy@cpc73037-sutt4-2-0-cust62.19-1.cable.virginm.net) |
| 10:30.15 | Tim_Toady | as for jumping there is goto, gotoif and some other options |
| 10:30.32 | Tim_Toady | in case you missed writing basic :P |
| 10:31.27 | guest9001 | http://pastebin.com/rjMw92c8 |
| 10:31.33 | guest9001 | With this in mind |
| 10:31.39 | guest9001 | it is the jump part I am not getting |
| 10:32.58 | Tim_Toady | _X will match only one digit extensions you probally need _X. or similar |
| 10:33.22 | guest9001 | well yeah, typo sorry. |
| 10:33.28 | guest9001 | But how do I send the call from the _+X |
| 10:33.33 | guest9001 | to the _.X |
| 10:33.40 | guest9001 | _X. |
| 10:35.50 | Tim_Toady | you can just do something like goto(${EXTEN:1}) |
| 10:35.57 | Tim_Toady | to strip the leading + |
| 10:36.14 | Tim_Toady | this works if you actually have somerthign that willl match ${EXTEN:1} |
| 10:36.20 | Tim_Toady | in your case you do |
| 10:38.27 | guest9001 | I have something that match 1? |
| 10:38.50 | guest9001 | exten => _X. ? I am with you if I had exten => 1001 |
| 10:38.52 | guest9001 | for an example |
| 10:39.05 | Tim_Toady | ${EXTEN:1} is the called extension stripped of the first digit, in tour case '+' |
| 10:39.10 | Tim_Toady | your* |
| 10:39.16 | guest9001 | oh I see |
| 10:39.24 | guest9001 | so I do not have to do any changes to callerid(num) |
| 10:39.27 | guest9001 | if I use this one |
| 10:40.14 | Tim_Toady | callerid is the ID of the phone that called, thats a different thing , we where talking about the called extension |
| 10:40.40 | *** join/#asterisk steelbrain (~steelbrai@103.255.5.56) |
| 10:41.02 | guest9001 | oh sorry, I am talking about the caller number |
| 10:41.11 | guest9001 | no.. I am not. |
| 10:41.14 | guest9001 | Sorry I am confused today :/ |
| 10:54.20 | guest9001 | Tim_Toady: thank you, I got it |
| 10:55.30 | Tim_Toady | :) |
| 11:02.19 | *** join/#asterisk Draecos (~Draecos@203-121-194-11.e-wire.net.au) |
| 11:06.56 | *** join/#asterisk kritzikratzi (~kritzikra@cpe90-146-150-86.liwest.at) |
| 11:12.04 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 11:12.37 | *** join/#asterisk juned (~juned@202.131.119.122) |
| 11:29.35 | *** join/#asterisk ThomasKeller (~Thomas@vmx-test.ethz.ch) |
| 11:37.53 | *** join/#asterisk italorossi (~Adium@187.60.66.11) |
| 11:50.04 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 11:54.53 | *** join/#asterisk C1ph3r (~C1ph3r@2804:7f3:280:5ae6:4847:aa80:55df:7eeb) |
| 12:15.26 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 12:16.14 | *** join/#asterisk guest9010 (~bounceman@185.32.9.250) |
| 12:17.35 | *** join/#asterisk gustopn (~gusto@mail.prolead-technologies.de) |
| 12:17.37 | gustopn | hi |
| 12:17.58 | gustopn | is there a standard that is being followed by the log output to /var/log/asterisk/messages ? |
| 12:18.20 | gustopn | is it like syslog log output, one log message per line, or can that message include a newline? |
| 12:19.11 | guest9010 | Tim_Toady: I was wrong, it did not work. I have attached some logs here http://pastebin.com/Z8PYWCvg |
| 12:19.15 | guest9010 | It never reaches STAGE 2 |
| 12:19.26 | guest9010 | Any ideas? |
| 12:20.31 | *** join/#asterisk Panther_Modern (~Panther_M@unaffiliated/panther-modern/x-6168176) |
| 12:21.28 | Tim_Toady | guest9010: try adding the extesion priority to goto, goto(${EXTEN:1}, 1) |
| 12:22.13 | guest9010 | Tim_Toady: that did it, why does it need the priority in this case? |
| 12:23.05 | Tim_Toady | 'core show application goto' in asterisk cli will answer that for u ;) |
| 12:28.24 | *** join/#asterisk juned (~juned@202.131.119.122) |
| 12:31.15 | *** join/#asterisk Zogot (~Adium@90-145-117-39.bbserv.nl) |
| 12:33.09 | *** join/#asterisk AviiNL (~AviiNL@185.21.52.255) |
| 12:36.30 | *** join/#asterisk ModFather (~ModFather@unaffiliated/modfather) |
| 12:53.42 | tparcina | I use HylaFAX with Asterisk (FreePBX). |
| 12:54.04 | tparcina | When I try to send fax SIP part of communication goes well. |
| 12:54.16 | *** join/#asterisk vinrock (~vin@unaffiliated/vinrock) |
| 12:54.52 | tparcina | But when RTP stream starts, my server sends packets with source IP address of wrong interface. |
| 12:55.17 | tparcina | I guess that's the reason why the other side can't hear anything from me. |
| 12:55.37 | tparcina | How to configure what source address should RTP stream use? |
| 12:56.21 | tparcina | And why it isn't, by default, the same one it's used in SIP part of communication? |
| 12:58.08 | *** join/#asterisk slav3_sergal (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 12:59.20 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 13:00.01 | *** join/#asterisk juned (~juned@202.131.119.122) |
| 13:07.57 | dan_j | tparcina: what version of asterisk are you using? |
| 13:19.24 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:25.40 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 13:27.54 | tparcina | dan_j: 11.17.1 |
| 13:35.32 | *** part/#asterisk juned (~juned@202.131.119.122) |
| 13:40.33 | *** join/#asterisk CeBe (~CeBe@a81-14-224-229.net-htp.de) |
| 13:41.31 | *** join/#asterisk Eyriqs (~Eyriqs@115.135.140.22) |
| 13:44.34 | Eyriqs | hello guys, i'm newbie and I desperately need help. I own a Linksys SPA400 and a synology server which already installed Asterisk GUI version 2.1.0. Question is how do I register spa400 on synology |
| 13:45.07 | Eyriqs | tq guys.. your help is appreciated :D |
| 13:46.33 | [TK]D-Fender | AsteriskGUI has not been supported for 5 years now |
| 13:46.44 | [TK]D-Fender | As for the rest, it's a SIP device like any otehr |
| 13:46.53 | [TK]D-Fender | make your peers just like any other device |
| 13:47.42 | Eyriqs | omg how back dated am I :O |
| 13:48.24 | Eyriqs | "make your peers just like any other device" does it means i have to registered it as a user? |
| 13:48.25 | [TK]D-Fender | Well you started the conversation mentioning a carbon-dated GUI ... and not even the actual Asterisk version. |
| 13:48.43 | [TK]D-Fender | Don't call it a "user" |
| 13:48.55 | Eyriqs | oh ok. I'm sorry then |
| 13:48.59 | [TK]D-Fender | A SIP peer is a SIP peer, is a SIP peer |
| 13:49.05 | Eyriqs | so how should i mention it |
| 13:49.08 | Eyriqs | ok SIP peer |
| 13:49.19 | Eyriqs | could you teach me how to setup a SIP Peer |
| 13:49.38 | [TK]D-Fender | There is functionally no difference to * between a typical SIP phone, a typical ITSP, or your SPA400 |
| 13:49.43 | [TK]D-Fender | ~book |
| 13:49.44 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 13:49.45 | [TK]D-Fender | ^^^^ |
| 13:50.12 | Eyriqs | wow cool script |
| 13:53.31 | Eyriqs | [TK]D-Fender, how do I configured SIP Peer on my synology? |
| 13:54.09 | *** join/#asterisk kritzikratzi (~kritzikra@193-80-14-60.adsl.highway.telekom.at) |
| 13:54.21 | [TK]D-Fender | Read the book, and google up some samples. Therer are several out there already |
| 13:54.45 | [TK]D-Fender | Also for similar items like the SPA-3012 which are largely the same |
| 13:55.38 | Eyriqs | [TK]D-Fender, as you said the GUI version has not been supported for 5 years. So no wonder there is limited source of information to configure this |
| 13:55.58 | Eyriqs | I have seeing a lot were made in ssh |
| 13:56.16 | Eyriqs | so i have no idea how should i start |
| 13:56.23 | [TK]D-Fender | [08:48][TK]D-FenderWell you started the conversation mentioning a carbon-dated GUI ... and not even the actual Asterisk version. <------- |
| 13:56.35 | [TK]D-Fender | You start by finding out what VERSION you even have on that thing |
| 13:56.45 | WIMPy | Installing a fresh system might be a possibility. |
| 13:57.17 | [TK]D-Fender | WIMPy: until you realize he's using an ancient shitty packacge on a NAS |
| 13:57.26 | [TK]D-Fender | Prepare for pain..... |
| 13:57.45 | Eyriqs | hi WIMPy, I had no problem with my Asterisk. |
| 13:57.55 | Eyriqs | everyhing is fine with external SIP |
| 13:58.11 | Eyriqs | but right now I would like to have my own FXO |
| 13:58.18 | Eyriqs | so I bought SPA400 |
| 13:58.33 | Eyriqs | I just dont know how to configure it |
| 14:00.24 | Eyriqs | thanks [TK]D-Fender |
| 14:00.33 | Eyriqs | i will try something else |
| 14:06.54 | *** join/#asterisk todwetsprock (~florian@p5099a594.dip0.t-ipconnect.de) |
| 14:07.49 | todwetsprock | hello! my DISA uses the wrong trunk -- i got 3 trunks thus 3 main telephone numbers. ideas ? |
| 14:08.31 | [TK]D-Fender | todwetsprock: It uses whatever context you told it to |
| 14:09.00 | [TK]D-Fender | todwetsprock: Go point it to one with extensions that use the peers you want it to use |
| 14:10.05 | todwetsprock | d-fender: i use extensions, those i can map to my trunks. now how to point my disa to a specific extension ? |
| 14:10.22 | [TK]D-Fender | .... |
| 14:10.31 | [TK]D-Fender | ~freepbx |
| 14:10.31 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 14:10.33 | [TK]D-Fender | ^^^ |
| 14:10.40 | WIMPy | DISA is for ENTERING an extension. |
| 14:10.46 | [TK]D-Fender | GUI's are not supported here |
| 14:11.47 | todwetsprock | 1) in freepbx noone answers ;) 2) id like to learn about asterisk 3) how do you know i use a gui ? :)) |
| 14:12.36 | todwetsprock | wimpy: thx, that information is helping in understanding my prob |
| 14:12.45 | WIMPy | Unless you learn the internals of the GUI first, knowing how to do things with Asterisk won't help you. -- Unless you scrap the GUI. |
| 14:13.02 | [TK]D-Fender | 1) yes they do. 2) Sure... go learn about Asterisk, but your question is a GUI questtion 3) I'm in #freepbx and you didn't ASK your quesiton in there so you don't complain about them not answering. You aren't in there ASKING. |
| 14:13.36 | todwetsprock | :( |
| 14:14.44 | todwetsprock | i dont even use freepbx but elastix. there, noone answers. and my freepbx plugin inside my elsastix doesnt do a thing waa |
| 14:15.07 | [TK]D-Fender | Stop using their garbage then. |
| 14:15.19 | todwetsprock | now i had plenty effort in configuring my asterisk, so i hope i can use asterisk to solve my prob thats why im here |
| 14:15.35 | [TK]D-Fender | "my freepbx plugin inside my elsastix doesnt do a thing waa" <- Doesn't do a thing? It does NOTHING? Totally broken? |
| 14:15.56 | [TK]D-Fender | If it's totally broken then how can DISA even be on your priority list? |
| 14:16.21 | todwetsprock | d-fender: you make me grin |
| 14:16.30 | *** join/#asterisk jasonwert (~wert@75-134-81-98.static.aldl.mi.charter.com) |
| 14:17.06 | [TK]D-Fender | You don't understand how the GUI works. |
| 14:17.07 | todwetsprock | mhh disa is working fine, but on the foreign phone the wrong number is shown. the number of trunk 3 :( |
| 14:17.11 | [TK]D-Fender | This is not an asterisk problem. |
| 14:17.16 | [TK]D-Fender | GUI's limit what you can do |
| 14:17.27 | [TK]D-Fender | You need to understand what it is meant to allow you to do/ |
| 14:18.15 | [TK]D-Fender | Along with the rest of the dialplan it generates |
| 14:18.30 | todwetsprock | mhhh inside /etc/asterisk/extensions_additional.conf theres plenty stuff about disa, so i thought maybe there to point my disa to an extension, point it on a trunk or just make it use the telephone number i like. |
| 14:19.21 | [TK]D-Fender | "plenty of stuff" says nothing |
| 14:19.23 | WIMPy | Possobly, but you have to do it the GUI way. Not an Asterisk question. |
| 14:19.25 | [TK]D-Fender | You don't go messing in there |
| 14:19.28 | [TK]D-Fender | the GUI generates that |
| 14:19.38 | [TK]D-Fender | So saying "there is stuff in there" is worthless\ |
| 14:19.55 | [TK]D-Fender | You have a dumb GUI. With like what ... 5 stupid fields in that GUI page for it? |
| 14:19.58 | [TK]D-Fender | THAT is what you can do |
| 14:20.23 | [TK]D-Fender | You don't "point it to an extension". That idea doesn't exist. |
| 14:20.52 | [TK]D-Fender | My initial phrasing was using proper Asterisk terms on the assumption you were actually configuring * yourself. You aren't |
| 14:21.18 | todwetsprock | wimpy: i get what u wanna say. now i had plenty effort in serving my gui with just editing the underlaying mysql for example |
| 14:21.54 | [TK]D-Fender | that MySQL is also not "asterisk" |
| 14:22.00 | [TK]D-Fender | that is the MySQL of the GUI itself |
| 14:22.24 | todwetsprock | ok. how to tell asterisk my disa to use the trunk id like ? |
| 14:22.28 | WIMPy | What ever way you do it, it still has to work with what the GUI configured. |
| 14:22.33 | [TK]D-Fender | Which is meaningless here. Again you have a simple GUI screen with a handful of fields on it. That is all you have to configure this with. |
| 14:23.03 | todwetsprock | ...but those fields are not inside my gui :(( |
| 14:23.34 | todwetsprock | <PROTECTED> |
| 14:23.34 | [TK]D-Fender | todwetsprock: You don't seem to be be showing anything useful here for even a hint.... |
| 14:23.40 | WIMPy | Well, GUIs will only allow you to use a certain subset of Asterisk. |
| 14:24.11 | [TK]D-Fender | senses a growing shift in density.... |
| 14:24.32 | todwetsprock | correct. so im here because i like to configure my asterisk in order my disa to use a certain telephone number |
| 14:24.36 | todwetsprock | is that possible ? |
| 14:24.43 | [TK]D-Fender | You aren't configuring Asterisk |
| 14:24.45 | [TK]D-Fender | Your GUI is |
| 14:24.50 | [TK]D-Fender | And you aren'tt cluing in |
| 14:24.52 | WIMPy | You need extensions for your trunks to select them, either by matching the destination or by means of prefixes. |
| 14:25.01 | todwetsprock | i do not want to use my gui. i have no gui. |
| 14:25.42 | [TK]D-Fender | You have Elastix. You've already confirmed it |
| 14:25.50 | [TK]D-Fender | lDon't give us some BS story now |
| 14:25.56 | [TK]D-Fender | You are asking about a stupid GUI and not even doing the one smart thing : SHOW US THE DAMN SCREEN SO WE CAN TELL YOU WHAT THAT LETS YOU DO. |
| 14:26.21 | todwetsprock | mmm as you said, i need my disa to point to an extension |
| 14:26.29 | [TK]D-Fender | You don't |
| 14:26.34 | [TK]D-Fender | DISA points to a CONTEXT. |
| 14:26.44 | *** join/#asterisk aross42 (~aross@dsl-eclipsefnd.storm.ca) |
| 14:26.48 | [TK]D-Fender | You clearly don't understand what DISA is in Asterisk let alone FreePBX |
| 14:26.52 | WIMPy | The user enters the extensions. |
| 14:26.53 | [TK]D-Fender | SCREENSHOT NOW. |
| 14:27.00 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 14:27.16 | [TK]D-Fender | This is turning into useless whining. |
| 14:27.30 | WIMPy | wonders if it's really DISA or just a call-through. |
| 14:29.02 | [TK]D-Fender | WIMPy: Yeah, FP actually calls DISA. What options his VERSION has ... we don't know because someone wasn't bright enough to even bother telling us what that is. So lets wait until he does the thing he should have done from the start: show us HIS screen so we can conrm what he can even do with what he has. |
| 14:30.52 | WIMPy | If it's a DISA function, you should be able to configure it like an extension (in the GUI sense). |
| 14:31.39 | [TK]D-Fender | WIMPy: It still just dumps them in a context. WHICH ONE is the question, and it's entirely dependent on what version his build is. |
| 14:31.49 | [TK]D-Fender | WIMPy: As to whether he has a choice at all or not. |
| 14:32.03 | *** join/#asterisk kritzikratzi (~kritzikra@193-80-14-60.adsl.highway.telekom.at) |
| 14:35.49 | todwetsprock | thank you very much so far. bye |
| 14:36.49 | [TK]D-Fender | And he doesn't even do the one thing that would help him. |
| 14:37.01 | [TK]D-Fender | \o/ |
| 14:49.22 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 15:00.56 | *** join/#asterisk sekil (~sekil@nat-73.net011.net) |
| 15:06.27 | DanQuinney | I going to buy [TK]D-Fender a pint if I ever see them in real life |
| 15:08.08 | *** join/#asterisk mub (~jub@107.1.220.2) |
| 15:08.52 | [TK]D-Fender | has apparently been upgraded to a "them" |
| 15:09.35 | [TK]D-Fender | I'm just good at calling a spade a spade... |
| 15:09.49 | WIMPy | Pluralis majestatis. |
| 15:10.33 | [TK]D-Fender | would buy WIMPy a pint |
| 15:11.06 | WIMPy | Oh, how did I get on that list? |
| 15:11.37 | WIMPy | thinks he might have to buy [TK]D-Fender a pint. |
| 15:12.23 | [TK]D-Fender | Sounds like a fair trade. I know who I'd definitely want at my table is all. |
| 15:15.31 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 15:18.29 | *** join/#asterisk kharwell (kharwell@nat/digium/x-sztfxvxmrgduxvcr) |
| 15:27.17 | *** join/#asterisk steelbrain (~steelbrai@103.255.5.56) |
| 15:27.23 | *** part/#asterisk steelbrain (~steelbrai@103.255.5.56) |
| 15:27.30 | *** join/#asterisk aneesiqbal (~steelbrai@103.255.5.56) |
| 15:28.08 | aneesiqbal | In chan_sip we used to have per-endpoint permit and deny, but all of that setting seems to have moved into a global acl for res_pjsip, any ideas how to achieve a similar ip-based auth behavior in PJSIP? |
| 15:28.32 | *** join/#asterisk TheTechStewart (~TheTechSt@overmind.thetechstewart.com) |
| 15:28.35 | file | permit and deny are not currently supported on a per-endpoint basis in PJSIP |
| 15:28.42 | file | it would have to be implemented |
| 15:30.16 | aneesiqbal | file: Is there a workaround? Any other method to achieve IP-based auth |
| 15:30.41 | file | PJSIP supports IP based matching, it is done using the type=identify section |
| 15:30.48 | [TK]D-Fender | I needs to be implemented... |
| 15:30.52 | [TK]D-Fender | It* |
| 15:32.17 | aneesiqbal | Yep, I've tried type=identify, Asterisk then knows that the request is from an endpoint. But then the auth section becomes a problem, if I remove the auth then every person would be able to login regardless of the IP, if I set it then the person with the correct IP will still have to provide a password |
| 15:32.54 | mub | Ever since we started using PJSIP I've seen hella log entries like "This feature is not available in PJSIP!" |
| 15:33.14 | mub | But I ignore them because no one is complaining |
| 15:33.37 | *** join/#asterisk kritzikratzi (~kritzikra@d86-32-54-38.cust.tele2.at) |
| 15:34.05 | aneesiqbal | There has to be a workaround for this, or I'll have to switch back to chan_sip which is a shame because I've already spent days trying to implement PJSIP |
| 15:39.34 | *** join/#asterisk Jesterboxboy (~Thunderbi@chello080109194026.3.graz.surfer.at) |
| 15:39.42 | *** join/#asterisk flujan (~flujan@200.160.115.22) |
| 15:39.59 | flujan | Hello, Is it possible to enable SIT detection on asterisk? |
| 15:44.23 | *** join/#asterisk ThomasKeller (~Thomas@vmx.ethz.ch) |
| 15:45.04 | aneesiqbal | Another question I have in mind is that can `realm` be an IP or does it really have to be a domain/ |
| 15:48.55 | aneesiqbal | cc [TK]D-Fender file mub |
| 15:49.26 | file | it can be whatever, doesn't even have to resolve |
| 15:49.55 | aneesiqbal | Does realm limit access by IP? |
| 15:50.12 | aneesiqbal | or it is a piece of decoration |
| 15:51.19 | file | it is used in SIP authentication |
| 15:51.24 | file | it does not limit access by IP |
| 15:51.24 | zaf | is it normal for Digium phones to register with their host_alternate server even though they can reach the primary just fine? |
| 16:04.00 | flujan | Hello, When I originate a Call using AMI and the company generate progress in band, the call is not connected to the context and AMD fails. How can I force the call to be connected to AMD and the context as soon as the telco generate progress in band? |
| 16:12.41 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-qzybhhzxuhnuaedd) |
| 16:24.40 | *** join/#asterisk AviiNL (~AviiNL@185.21.52.255) |
| 16:32.14 | *** join/#asterisk nunopereira (~nuno.pere@alpes.nortenet.pt) |
| 16:43.02 | *** join/#asterisk nunopereira (~nuno.pere@alpes.nortenet.pt) |
| 16:53.33 | *** join/#asterisk jgowdy (bt@hur.ruhr) |
| 16:54.09 | jgowdy | If I have direct media / can reinvite set to true and then call MixMonitor to record a call, will it reinvite Asterisk back into the middle of the audio? |
| 16:56.07 | [TK]D-Fender | Call it how? |
| 16:56.17 | jgowdy | DialPlan or AsyncAGI |
| 16:56.46 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-86-39.hsd1.or.comcast.net) |
| 16:57.39 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 16:58.41 | [TK]D-Fender | If you're in the dialplan then you haven't reinvited ANYWHERE yet |
| 16:58.52 | [TK]D-Fender | and when a call is aplaced it will not even TRY to reinvite... because it ahs to record |
| 17:03.57 | jgowdy | I mean I'm in the dial plan throughout the call. What I'm trying to figure out is if I Dial or Bridge and Asterisk negotiates direct media, then unbridge via ChannelRedirect, then enable MixMonitor and bridge again, will Asterisk not use direct media due to the MixMonitor running |
| 17:05.03 | *** join/#asterisk timotheus1 (~roost@unaffiliated/roost) |
| 17:10.03 | [TK]D-Fender | clearly not |
| 17:10.07 | [TK]D-Fender | it shouldn't be allowed |
| 17:10.12 | [TK]D-Fender | can't record what isn't there |
| 17:10.58 | jgowdy | Right, but it could obviously trigger a reinvite back to non-direct media when something like MixMonitor runs. It's not impossible. If it doesn't do that, cool, good to know. |
| 17:12.31 | [TK]D-Fender | If you have monitor on the channel it shouldn't allow further reinvites on anything "bridged" |
| 17:18.07 | *** join/#asterisk mhterres (~marcelo@ltsp4.propus.com.br) |
| 17:18.24 | mhterres | hello, anybody here can help me with AOR and pjsip? |
| 17:18.35 | mhterres | I'm migration SIP to PJSIP |
| 17:18.56 | mhterres | And when I try to register my extension a I received this message |
| 17:18.59 | *** join/#asterisk spicyramen (~Adium@216.239.45.89) |
| 17:19.03 | *** join/#asterisk jvhester (Adium@nat/digium/x-ioumcteutwqwsmbs) |
| 17:19.15 | mhterres | *CLI> [Jan 11 15:11:27] WARNING[14462]: res_pjsip_registrar.c:678 registrar_on_rx_request: AOR '10' not found for endpoint '10' |
| 17:19.25 | mhterres | what am I doing wrong? :-/ |
| 17:19.37 | mhterres | I'm trying to use dynamic and not static endpoints |
| 17:19.55 | mhterres | I want to register the account in 2 different softphones |
| 17:21.32 | *** join/#asterisk vader- (~Adium@50.232.174.194) |
| 17:23.39 | *** join/#asterisk sparetire_ (~sparetire@unaffiliated/sparetire) |
| 17:23.42 | [TK]D-Fender | Seems to clearly say that an AOR you said it should look at.... does not exist. |
| 17:23.49 | [TK]D-Fender | Show us that you made it properly |
| 17:24.35 | mhterres | I put everything in postgres and I'm accessing it via ODBC |
| 17:24.45 | [TK]D-Fender | SHOW US |
| 17:24.52 | mhterres | Off course |
| 17:25.00 | mhterres | I'm using extension 10 to test it |
| 17:25.01 | mhterres | pjsip show aors |
| 17:25.01 | mhterres | <PROTECTED> |
| 17:25.01 | mhterres | <PROTECTED> |
| 17:25.01 | mhterres | <PROTECTED> |
| 17:25.01 | mhterres | <PROTECTED> |
| 17:25.17 | mhterres | *CLI> pjsip show endpoints |
| 17:25.17 | mhterres | <PROTECTED> |
| 17:25.17 | mhterres | <PROTECTED> |
| 17:25.17 | mhterres | <PROTECTED> |
| 17:25.17 | mhterres | <PROTECTED> |
| 17:25.19 | mhterres | <PROTECTED> |
| 17:25.21 | mhterres | <PROTECTED> |
| 17:25.23 | mhterres | <PROTECTED> |
| 17:25.25 | mhterres | <PROTECTED> |
| 17:25.27 | mhterres | <PROTECTED> |
| 17:25.29 | mhterres | <PROTECTED> |
| 17:25.31 | mhterres | <PROTECTED> |
| 17:25.33 | mhterres | <PROTECTED> |
| 17:25.35 | mhterres | <PROTECTED> |
| 17:25.37 | mhterres | <PROTECTED> |
| 17:25.39 | mhterres | *CLI> pjsip show auths |
| 17:25.41 | mhterres | <PROTECTED> |
| 17:25.43 | mhterres | <PROTECTED> |
| 17:25.47 | mhterres | <PROTECTED> |
| 17:25.49 | mhterres | Did I miss something ? |
| 17:25.55 | mub | Wwaait wait pastebin it or something |
| 17:26.04 | mub | My IRC window is really small |
| 17:26.07 | mhterres | sorry |
| 17:26.12 | [TK]D-Fender | PASTEBIN <---------------- |
| 17:26.19 | WIMPy | Don't you have auto-ignore? |
| 17:26.23 | WIMPy | Worked for me :-) |
| 17:26.25 | mhterres | I didn't paste bin it cause it's small, but I'll do it now |
| 17:26.45 | [TK]D-Fender | That was not "small" |
| 17:26.56 | [TK]D-Fender | <PROTECTED> |
| 17:27.08 | [TK]D-Fender | that clearly is not what you told it to look for |
| 17:27.29 | mhterres | http://pastebin.com/7yG7fg2t |
| 17:27.51 | [TK]D-Fender | : AOR '10' not found for endpoint '10' |
| 17:28.01 | [TK]D-Fender | that is saying there is no AOR = "10" |
| 17:28.04 | [TK]D-Fender | Which there isn't |
| 17:28.14 | [TK]D-Fender | you have one called "aor10" |
| 17:28.22 | *** join/#asterisk spicyramen (~Adium@216.239.45.89) |
| 17:28.37 | *** join/#asterisk superscrat (asanders@nat/digium/x-wsgrkmnhadecctnk) |
| 17:28.38 | mhterres | but in DB in aors column I put aor10 |
| 17:29.11 | [TK]D-Fender | Doesn't look like it's reloaded that fact. |
| 17:29.27 | mhterres | LOG: statement: select id,max_contacts from ps_aors where id='aor10'; |
| 17:29.28 | mhterres | <PROTECTED> |
| 17:29.28 | mhterres | -------+-------------- |
| 17:29.28 | mhterres | <PROTECTED> |
| 17:30.22 | mhterres | LOG: statement: select id,transport,aors,auth,callerid from ps_endpoints where id='10'; |
| 17:30.22 | mhterres | <PROTECTED> |
| 17:30.22 | mhterres | ----+---------------+-------+--------+-------------- |
| 17:30.22 | mhterres | <PROTECTED> |
| 17:30.46 | mhterres | Is it wrong? |
| 17:30.57 | file | when a device registers it specifies the AOR it wants to use, unless the device is specifying 'aor10' then it won't use it - it looks like it is specifying '10' instead |
| 17:32.14 | mhterres | well, I created an aor10 and it's not working anyway |
| 17:32.38 | file | because the device attempting to register is saying 'I want to use an AOR named 10' |
| 17:32.43 | file | not 'I want to use an AOR named aor10' |
| 17:32.50 | mhterres | http://pastebin.com/b5NsYTtm |
| 17:33.00 | mhterres | so, now should work, correct ? |
| 17:33.11 | file | if you allow it on the endpoint, yes |
| 17:33.28 | mhterres | let me try this |
| 17:33.39 | *** part/#asterisk flujan (~flujan@200.160.115.22) |
| 17:34.14 | mhterres | thanks everybody |
| 17:34.17 | mhterres | It's working now |
| 17:37.17 | *** join/#asterisk Tim_Toady (~fuzzy@snf-33276.vm.okeanos.grnet.gr) |
| 17:40.31 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 17:49.23 | *** join/#asterisk italorossi (~Adium@177.65.201.177) |
| 17:53.32 | mhterres | anybody here is using pjsip realtime ? |
| 17:53.52 | mhterres | When I register my account, I received the following error in console |
| 17:54.04 | mhterres | *CLI> [Jan 11 15:53:07] WARNING[8853]: res_odbc.c:649 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: HY000: ERROR: invalid input syntax for integer: "3.000000" at character 173; |
| 17:54.04 | mhterres | Error while executing the query (102) |
| 17:54.13 | mhterres | The problema here is '3.00000' |
| 17:54.21 | mhterres | If I use just '3' it works |
| 17:55.05 | mhterres | I noticed that when I got the queries in postgres |
| 17:59.06 | *** join/#asterisk spicyramen (~Adium@216.239.45.89) |
| 18:07.09 | [TK]D-Fender | Well you can't have decimals for an integer now can you? |
| 18:07.22 | [TK]D-Fender | So the ".000000" isn't playing nice |
| 18:28.41 | mhterres | yes, I reported a bug |
| 18:34.48 | *** join/#asterisk sekil (~Ognjen@adsl-82-117-222-227.dynamic.sbb.rs) |
| 18:35.42 | *** join/#asterisk boson (~boson@2605:a000:fc44:5740:a800:acff:fe10:813b) |
| 18:47.04 | mhterres | Variable qualify_timeout in Asterisk is double and table column is integer. |
| 18:47.04 | mhterres | I changed the db column to double precision and now it is working. |
| 18:47.12 | mhterres | I reported in the issue |
| 18:48.45 | *** join/#asterisk steelbrain (~steelbrai@103.255.5.56) |
| 18:48.47 | *** part/#asterisk steelbrain (~steelbrai@103.255.5.56) |
| 18:48.56 | *** join/#asterisk aneesiqbal (~steelbrai@103.255.5.56) |
| 18:49.11 | aneesiqbal | Okay I've migrated back to chan_sip but now I'm hitting a new problem lol |
| 18:49.48 | aneesiqbal | Any ideas how do I get chan_sip to treat all requests from a certain ip as a certain user? PJSIP's identify module used to do it |
| 18:51.50 | [TK]D-Fender | Make a peer for it |
| 18:54.27 | aneesiqbal | I have, http://pastebin.com/raw/E8vSzDvM |
| 18:55.01 | [TK]D-Fender | that does not say "peer" |
| 18:55.15 | aneesiqbal | do I have to name it "peer" to make IP-based auth work? |
| 18:55.38 | [TK]D-Fender | You don't seem to understand what the "type" field means |
| 18:55.56 | aneesiqbal | For clarification, the account will be calling my asterisk instance with multiple random usernames |
| 18:56.08 | [TK]D-Fender | peer = auth by IP |
| 18:56.16 | [TK]D-Fender | user = auth by username |
| 18:56.22 | [TK]D-Fender | friend = BOTH |
| 18:56.39 | aneesiqbal | Oh right, some wiki auth said peers can only do outbound calls and not inbound calls |
| 18:56.51 | [TK]D-Fender | Next if you want to allow calls from that IP to match... you shouldn't CHALLENGE their calls |
| 18:56.59 | [TK]D-Fender | that is WRONG |
| 18:57.16 | aneesiqbal | Please advise a better way |
| 18:58.44 | [TK]D-Fender | insecure=port,invite |
| 18:58.47 | [TK]D-Fender | type=peer |
| 18:59.35 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-qfftntklpjvvuxvo) |
| 18:59.35 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 19:00.20 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 19:01.58 | aneesiqbal | That's a good idea! |
| 19:03.22 | aneesiqbal | Thanks [TK]D-Fender! |
| 19:07.45 | *** join/#asterisk superscrat (asanders@nat/digium/x-oqbzpqtdjuzbmtjw) |
| 19:15.05 | DanQuinney | Sorry [TK]D-Fender, I didn't know if you are a him or a her. Although you can have a pint either way. |
| 19:16.49 | [TK]D-Fender | "Not sure if you're a boy or a girl" - David Bowie (1947-2016) |
| 19:19.34 | *** join/#asterisk saratogga (~saratogga@wsip-98-174-232-77.ph.ph.cox.net) |
| 19:24.07 | sekil | hot tramp I love you so.. |
| 19:27.43 | *** join/#asterisk aneesiqbal (~steelbrai@103.255.7.5) |
| 19:29.25 | aneesiqbal | That fixes one issue of password for the caller, anyone got a possible fix on making chan_sip recognize random usernames as a particular peer based on their IP? |
| 19:53.11 | [TK]D-Fender | The whole point of peer is that username doesn't matter. |
| 19:53.23 | [TK]D-Fender | Not sure what partt of "identifies by IP" is unclear |
| 19:54.30 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-172.cust.bezeqint.net) |
| 20:00.46 | *** join/#asterisk aness (~aness@2a02:fe0:c310:4830:3409:f37b:9c5f:c7a3) |
| 20:02.57 | *** join/#asterisk thiagoc (~thiagoc@unaffiliated/thiagoc) |
| 20:03.21 | thiagoc | hi, anyone have experience with chan_dongle? |
| 20:04.38 | thiagoc | i'm using a Sipolar 16 ports but my customer complains about the quality of calls |
| 20:04.47 | *** part/#asterisk saratogga (~saratogga@wsip-98-174-232-77.ph.ph.cox.net) |
| 20:05.58 | thiagoc | I know it's not a enterprise solution but I know some people use this without problems |
| 20:06.46 | [TK]D-Fender | You've got BT quality ... then cell quality ... then whatever the other end is using. |
| 20:13.14 | *** join/#asterisk Synthase_ (uid63346@gateway/web/irccloud.com/x-flxpclrfsbdgrukd) |
| 20:27.07 | thiagoc | also I'm getting -99dBm, is this acceptable? |
| 20:31.41 | *** join/#asterisk jasonwert (~wert@75-134-81-98.static.aldl.mi.charter.com) |
| 20:33.35 | *** join/#asterisk kritzikratzi (~kritzikra@cpe90-146-150-86.liwest.at) |
| 20:38.14 | *** join/#asterisk superscrat (asanders@nat/digium/x-tvovbcrbgfwjmzxo) |
| 20:39.34 | *** join/#asterisk lgaetz (~lgaetz@blk-252-35-203.eastlink.ca) |
| 20:41.15 | *** join/#asterisk aneesiqbal (~steelbrai@103.255.5.36) |
| 20:41.32 | *** join/#asterisk spicyramen (~Adium@216.239.45.89) |
| 20:41.51 | lgaetz | Anyone have a definitive answer whether Asterisk is multi-threaded? Or is it a question of whether it's SIP/PJSIP which is multithreaded |
| 20:42.00 | lgaetz | FreePBX support question from a client |
| 20:42.20 | lgaetz | ^^@mjordan |
| 21:25.59 | mub | Yes it has to do with the driver itself |
| 21:26.22 | mub | :( |
| 21:27.06 | mub | I don't understand people who join, ask a question, then leave 5 minutes later |
| 21:27.15 | mub | Except when I do it |
| 21:27.45 | Kobaz | [2016-01-11 16:19:37.732] -- [C-0000009a] <SIP/pbx-sip-t1-gw1-0000008a> x=0, open writing: /var/spool/asterisk/voicemail/internal/219/tmp/ROHt90 format: wav, 0x861877c |
| 21:27.45 | Kobaz | [2016-01-11 16:19:37.732] -- [C-0000009a] <SIP/pbx-sip-t1-gw1-0000008a> was put on hold by its bridged peer: (None) |
| 21:27.47 | Kobaz | that seems strange |
| 21:27.53 | Kobaz | put on hold while leaving voicemail? |
| 21:30.06 | *** join/#asterisk scjones (~01scjones@ip72-193-183-206.lv.lv.cox.net) |
| 21:31.04 | mub | Kobaz: My asterisk box does not do that, so yeah that's strange |
| 21:33.51 | mub | Kobaz: Call them and ask why they would put a vm box on hold |
| 21:34.24 | Kobaz | call who? |
| 21:34.28 | Kobaz | "them" |
| 21:36.58 | mub | They killed Kenny! |
| 21:44.07 | *** join/#asterisk Demon_VoIP (~demon@ip253.net222.n37.ru) |
| 21:45.27 | *** join/#asterisk war9407 (war@static-72-73-18-14.clppva.fios.verizon.net) |
| 21:47.32 | *** join/#asterisk rrittgarn (~rrittgarn@75-150-221-196-Illinois.hfc.comcastbusiness.net) |
| 22:04.32 | *** join/#asterisk rogerh (~rogerh@ip202-37-70-18.satlan.co.nz) |
| 22:07.40 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:20.26 | *** join/#asterisk lgaetz_ (~lgaetz@blk-252-35-203.eastlink.ca) |
| 22:20.42 | *** part/#asterisk lgaetz_ (~lgaetz@blk-252-35-203.eastlink.ca) |
| 22:37.53 | *** join/#asterisk C1ph3r (~C1ph3r@179.182.203.138) |
| 22:50.15 | *** join/#asterisk jonno11 (~Jon@88.202.172.162) |
| 22:56.05 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 23:10.46 | *** join/#asterisk pankid (~quassel@mail.nschoonover.com) |
| 23:11.12 | pankid | anyone know of a phone that allows provisioning to be downloaded via sftp? would be cool to have a phone that just uses an ssh connection to download configs. |
| 23:30.39 | dan_j | Kobaz: Sometimes when the other side hangs up, I get put on hold. Not figured out why yet. |
| 23:30.59 | dan_j | Hold music lasts for about 4 seconds and then asterisk hangs up my channel. |
| 23:31.26 | WIMPy | Sounds like the analog days. |
| 23:31.43 | dan_j | With me it's sip to sip. no analog involved |
| 23:32.16 | WIMPy | So I guess it's doing a good analog emulation *eg* |
| 23:37.47 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-iczpcumfrfawcpfo) |
| 23:37.47 | *** mode/#asterisk [+o newtonr] by ChanServ |
