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| 02:08.08 | KNERD | Asterisk is giving me a warning " The application delimiter is now the comma, not the pipe. Did you forget to convert your dialplan? " |
| 02:08.42 | KNERD | However, I am not using it as a "delimter" but as an operator |
| 02:09.22 | KNERD | Maybe I am doing the function wrong? |
| 02:09.31 | KNERD | "(DEVSTATE(1000) = UNKNOWN | NOT_INUSE | INUSE | BUSY | INVALID | RINGING | RINGINUSE | ONHOLD ) |
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| 02:26.17 | m0sphere | anyone here happen to have a Cisco SPA122 ATA? |
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| 02:36.42 | Narkov2 | When a Local channel makes a call using the Dial command, is it possible to retrieve the HANGUPSTATUS of the Dial and not just the Local channel? |
| 02:48.47 | WIMPy | KNERD: What is that meant to do? You have to pick one, not all. |
| 02:49.24 | WIMPy | Narkov2: that should be passed on AFAIK. |
| 02:53.46 | Narkov2 | WIMPy: I can only seem to get the Local channels HANGUPSTATUS |
| 02:53.47 | KNERD | no |
| 02:53.49 | KNERD | i nee ALL of them |
| 02:57.36 | WIMPy | Narkov2: I was sure that would be the same as what it received. I once patched it to ensure that it would handle the 'answered elsewhere' case correctly. |
| 02:57.46 | WIMPy | KNERD: You can only have one at a time. |
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| 02:59.56 | Narkov2 | WIMPy: My apologies. It's the HANGUPCAUSE that isn't propergating. |
| 03:01.13 | WIMPy | That's one of the possible causes. Just that it also has (or had?) an additional flag. |
| 03:01.41 | KNERD | WIMPy: but I need to know if ti falls into any of those statuses |
| 03:02.17 | WIMPy | KNERD: What exactely are you trying to do? |
| 03:02.26 | KNERD | then why is UNREACHABLE not an ption? |
| 03:02.38 | KNERD | I need to know if any phoen is UNREACHABLE |
| 03:02.44 | KNERD | then it so, call a cell phone |
| 03:04.08 | WIMPy | Use SIPPEER(status) |
| 03:06.17 | WIMPy | Or just try to call it and if it fails... |
| 03:07.24 | KNERD | WIMPy: thanks..i will give that a try |
| 03:12.36 | KNERD | WIMPy: this look like proper syntax? GotoIf(SIPPEER(1000|status) = UNREACHABLE ?:phone2) |
| 03:13.27 | WIMPy | Nope. |
| 03:13.39 | WIMPy | Maybe it was up to 1.2. |
| 03:13.46 | WIMPy | 'core show application GotoIf' |
| 03:13.56 | KNERD | http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-APP-E-51.html |
| 03:15.17 | WIMPy | That is obviousely pre 1.4 syntax. The pipe has gone. We use commas now. |
| 03:18.41 | KNERD | oh...okay |
| 03:24.07 | KNERD | WIMPy: http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#Database_id243940 |
| 03:24.17 | KNERD | SIPPER is not mentioned anywhere |
| 03:24.29 | KNERD | sorry.. SIPPEER |
| 03:24.54 | WIMPy | So use 'core show function SIPPEER'. |
| 03:25.47 | KNERD | https://wiki.asterisk.org/wiki/display/AST/Function_SIPPEER?src=search |
| 03:25.50 | KNERD | oh..thanks |
| 03:26.09 | WIMPy | Or the wiki, yes. |
| 03:26.43 | KNERD | still notgood info |
| 03:26.58 | KNERD | in the status, what are the vauilable outputs?? |
| 03:27.04 | KNERD | available |
| 03:28.46 | KNERD | WIMPy: no way to execute it from the CLI? |
| 03:29.10 | WIMPy | What? A dialplan function? No. |
| 03:29.51 | KNERD | then whow am I suppose to know all the available outputs it gives? |
| 03:30.07 | WIMPy | Read the source. |
| 03:30.25 | WIMPy | Or ask for better documentation. |
| 03:32.45 | KNERD | thanks |
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| 04:22.42 | KNERD | WIMPy: found it |
| 04:23.25 | KNERD | UNREACHABLE is a valid choice |
| 04:36.12 | WIMPy | ok |
| 04:38.53 | KNERD | but still nort working..gosh |
| 04:39.23 | KNERD | WIMPy: dont mind takigng a peek at this script i got? |
| 04:40.26 | KNERD | http://pastebin.com/Auyve5MB |
| 04:42.46 | [TK]D-Fender | perhaps you should use an actual expression.... |
| 04:43.29 | WIMPy | And double quotes are literal. If you use them, you need to use them on both sides. |
| 04:44.50 | KNERD | double qtotes remove |
| 04:45.07 | KNERD | ANd what do you mean by "actual expression" [TK]D-Fender: |
| 04:45.23 | [TK]D-Fender | You know Asterisk Expressions? USE them. |
| 04:46.26 | KNERD | I am using what is shown here http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-CHP-6-SECT-3.html |
| 04:46.36 | KNERD | I dont know how I can get "more asterisk" than that |
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| 04:47.37 | [TK]D-Fender | Yes, and your code is missing crucial bits you can see in the book right there |
| 04:47.39 | KNERD | Conditional Branching Prev Chapter 10. Deeper into the Dialplan Conditional Branching' |
| 04:48.18 | KNERD | GotoIf(expression?destination1:destination2) |
| 04:48.19 | WIMPy | There *might* be a page about expressions in the wiki. |
| 04:48.35 | [TK]D-Fender | KNERD: It's your lack of having a valid expression... |
| 04:48.51 | KNERD | http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-CHP-6-SECT-1.html#asterisk-CHP-6-SECT-1.1 |
| 04:49.00 | [TK]D-Fender | KNERD: Looks at theirs.. look at yours.... notice the glaring difference.... |
| 04:50.17 | KNERD | ules you mean this" same =>" then no |
| 04:50.36 | [TK]D-Fender | KNERD: Look harder..... |
| 04:50.56 | KNERD | the brackets? [] |
| 04:50.59 | [TK]D-Fender | YES |
| 04:51.06 | [TK]D-Fender | $[] <- EXPRESSION |
| 04:54.05 | KNERD | okay..hchanging it |
| 04:54.36 | KNERD | OK in single quote or no quotes [TK]D-Fender: |
| 04:55.09 | [TK]D-Fender | quote have to be on both sides and only used if you risk having one side evaluate as blank |
| 04:57.01 | KNERD | what do you mena by "both sides"? of what? |
| 04:57.29 | WIMPy | The expression. |
| 04:57.45 | WIMPy | Quotes are nothign special. Just like any other character. |
| 04:59.34 | KNERD | Still unsure as non of the exampels have any quotes |
| 05:00.17 | KNERD | when I put quotes in there in :OK" to em that means the excact string, OK |
| 05:07.38 | KNERD | WIMPy: [TK]D-Fender: okay..it is workin gnow..thnkas for the advice |
| 05:10.04 | KNERD | The thing is on the last line, IO really don't want to hang up, but rather retuern |
| 05:17.59 | Penguin | "OK" is totally not the same as OK |
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| 05:23.24 | KNERD | wel it work snow...i did no quotes |
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| 06:39.18 | D30 | hi all, |
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| 09:48.41 | dan_j | Hi, I'm looking into load balance and high availability. Am I correct in saying that if opensers is used to load balance between two asterisk servers, that can break call queues if agents are registered on one asterisk server, and the callers have ended up on another asterisk server? |
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| 10:10.55 | NiugeS | hi all.. anyone awake? |
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| 10:15.42 | Chainsaw | We are all asleep at this time. |
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| 10:30.40 | NiugeS | Okay - a bit of a newbie so please be kind! Had someone set us up with an Elastix setup with Cisco phones ... Is there a way to have more than two people added to a conference call? (user dials dials to receiver 1, and once on, then conferences receiver 2, we need receiver 3).. there is not the option of them calling into us.. |
| 10:33.18 | Chainsaw | So you are using the Conference feature on the Cisco phones themselves, instead of using the Asterisk/Elastix conference bridges? |
| 10:33.37 | Chainsaw | I suspect you'll have to transfer people into a bridge, rather then using the conference key. |
| 10:34.48 | NiugeS | Chainsaw yes.. on eht phone itself. |
| 10:35.08 | NiugeS | is there a link to more info on setting up a bridge etc? |
| 10:35.31 | Chainsaw | NiugeS: You'd have to see what the Elastix way of doing it is; they've integrated Asterisk into a bigger work. |
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| 10:36.09 | Chainsaw | NiugeS: There is an #elastix channel that may be better suited to the question, now that you know what you're after. |
| 10:36.27 | Chainsaw | NiugeS: If you find out how to use the conference bridge instead of the phone conference button, your limitations will go away. I can guarantee you that. |
| 10:37.45 | NiugeS | Chainsaw i'm in the channel however it appears no one is awake ;).. i will look into the bridge side of things to see if I am able to find anything |
| 10:37.46 | NiugeS | thank you |
| 10:38.03 | Chainsaw | NiugeS: Any time. |
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| 12:54.07 | Ice_Strike | Anyone here work at the call centre? How do you guys power up 10 computers with phones and monitors? I am currently using extention and chained with other extentions.. 30 plugs! |
| 12:54.55 | leifmadsen | PoE |
| 12:55.01 | Ice_Strike | The phones currently making buzzing noise when not on the calls and during the calls. |
| 12:55.03 | WIMPy | In a tent? |
| 12:56.10 | WIMPy | Dado trunking |
| 12:56.46 | Ice_Strike | 30 plugs of Dado trunking? |
| 12:57.05 | WIMPy | of? in. |
| 12:57.25 | Ice_Strike | We have ethernet sockets in Dado trunking |
| 12:57.34 | Ice_Strike | but limited power sockets. |
| 12:58.03 | WIMPy | Put more in. |
| 12:58.57 | Greenlight | Buzzing noise is usually a bad earth somewhere |
| 13:01.15 | SpeedEvil | Or a trapped bee. |
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| 13:02.07 | [TK]D-Fender | Perhaps you should ask in #MaybeAnElectricianIsInOrder |
| 13:02.44 | Ice_Strike | http://s28.postimg.org/lho63iwn1/desk.png |
| 13:02.49 | Ice_Strike | It look something like that |
| 13:03.50 | Ice_Strike | Greenlight bad earth via? |
| 13:03.54 | [TK]D-Fender | get your phones on a power-conditioner |
| 13:04.09 | Ice_Strike | Is it not because I am using a lot of extentions as chained that cause buzzing noise? |
| 13:05.00 | Greenlight | That's unlikely to be the cause... |
| 13:05.35 | Ice_Strike | [TK]D-Fender What is power-conditioner? I use Power Surge Extensions |
| 13:06.42 | Greenlight | Power conditionor will clean dirty power supply |
| 13:06.56 | Greenlight | It's completely differnt (and more expensive) than a surge protector |
| 13:09.00 | Ice_Strike | is it this one: http://www.amazon.co.uk/LINDY-Mains-Conditioner-Power-Strip/dp/B00289GSC0/ref=sr_1_3?ie=UTF8&qid=1394716104&sr=8-3&keywords=power+conditioner |
| 13:09.15 | WIMPy | With switching PSUs it's rather unlikely that any condition on the high voltage side would cause anything. |
| 13:09.34 | WIMPy | More likely is bad shielding and radiating monitors or something. |
| 13:12.04 | Chainsaw | Ice_Strike: Your best bet is still to power the phones with PoE, leaving the sockets free for the PCs. |
| 13:12.27 | Chainsaw | Ice_Strike: I concur with Greenlight that it sounds like a ground loop. |
| 13:12.44 | WIMPy | Or build adapters to power the phones from the PCs. Makes them a little less expensive. |
| 13:13.05 | Chainsaw | WIMPy: And there is the ground loop again, congratulations. |
| 13:13.44 | WIMPy | Use cheaper ethernet cables without ground connection then. |
| 13:13.46 | Chainsaw | WIMPy: A cute little PoE ProCurve doesn't cost much, particularly if you don't need it to be fanless. |
| 13:14.02 | Chainsaw | WIMPy: Redo the DADO trunking and cabling end-to-end? |
| 13:14.11 | Chainsaw | WIMPy: *cash register sound* |
| 13:14.43 | Chainsaw | WIMPy: Now you can just get an Extreme X460 and still be cheaper. |
| 13:14.56 | WIMPy | Actually I'm not sure I have seen any ground connection on voip phones. They all had punshielded ethernet ports so far. |
| 13:15.27 | [TK]D-Fender | Impenetrable to jokes? |
| 13:15.58 | WIMPy | oops |
| 13:18.27 | WIMPy | Anyway. For a ground loop you need more than one earth connection. Seems like a bit of a challenge on a device that doesn't even have a single ground connection. |
| 13:19.07 | Greenlight | The switch is usually grouned. |
| 13:19.07 | Chainsaw | WIMPy: You've already seen that the power supply has no ground connection? |
| 13:19.23 | Chainsaw | WIMPy: You've already ruled out shielded CAT5/6 runs? |
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| 13:19.41 | Chainsaw | WIMPy: That's quick going. I'll leave it with you. |
| 13:19.44 | michael_work | hello |
| 13:19.48 | Chainsaw | Hi Michael. |
| 13:20.18 | michael_work | my google kungfu and search over jira failed me |
| 13:20.25 | WIMPy | I just wrote that I haven't seen a shielded ethernet port on a phone, yet. And the PSUs don't have a ground connection, either. Or if, certainly not on the secondary side. |
| 13:20.26 | Chainsaw | Happens to the best of us. |
| 13:20.31 | Chainsaw | What's the question Google failed to answer? |
| 13:20.33 | michael_work | is there problem with IAX2 and asterisk 11 ? |
| 13:20.40 | michael_work | it seems it's not working anymore |
| 13:20.43 | Chainsaw | michael_work: No, Asterisk 11 is perfect. |
| 13:20.48 | michael_work | i see the packets get to server but asterisk is silent |
| 13:20.59 | Chainsaw | WIMPy: Polycom SoundPoint IP670. |
| 13:21.06 | michael_work | Chainsaw, i know it's perfect, but the question is if IAX@ works :) |
| 13:21.28 | michael_work | Chainsaw, what's the version on asterisk you can confirm it works |
| 13:21.53 | Chainsaw | michael_work: 11.8.1 |
| 13:22.09 | michael_work | hmm |
| 13:22.19 | michael_work | i have Asterisk 11.5.0 |
| 13:22.30 | michael_work | Asterisk 11.6-cert1 |
| 13:22.35 | michael_work | and bunch of 1.8 |
| 13:22.40 | michael_work | 1.8 works to each other |
| 13:22.53 | michael_work | they fail to 11 and 11 between eachother fail |
| 13:22.54 | Chainsaw | michael_work: 11.5.0 is subject to several security issues by now. I hope it's not talking to the big bad internet. |
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| 13:23.14 | michael_work | Chainsaw, any chance for url? |
| 13:23.24 | Chainsaw | michael_work: What URL do you require? |
| 13:23.25 | michael_work | for more info so i can scary my bosses |
| 13:23.43 | michael_work | i never require i ask for :) |
| 13:24.06 | Chainsaw | michael_work: http://www.asterisk.org/downloads/security-advisories |
| 13:24.31 | Chainsaw | michael_work: Disregard the top two as they only apply to Asterisk 12. Most of what follows below that is relevant to 11.5.0; just read the PDFs and stop when you see 11.5 is the secure version. |
| 13:24.40 | Chainsaw | michael_work: Then print them all out for your scary book of doom. |
| 13:25.56 | michael_work | :) |
| 13:25.58 | michael_work | thanks |
| 13:26.05 | michael_work | now back to IAX2 |
| 13:26.18 | michael_work | it's wierd i see it's comming to server but i do not see any response |
| 13:26.25 | michael_work | and debug is not working |
| 13:26.50 | [TK]D-Fender | prove your firewall dump, prove that the module is loaded. check your peers. enabel debug. show everything |
| 13:26.58 | michael_work | i mean it doesn't print anything |
| 13:27.00 | michael_work | no info |
| 13:27.04 | [TK]D-Fender | include tcpdump on the side as well |
| 13:27.09 | michael_work | like asterisk is not even listening to it |
| 13:27.23 | michael_work | ok |
| 13:27.32 | michael_work | i have tcpdump from yesterday |
| 13:27.44 | michael_work | iptables inludes any to any |
| 13:27.50 | michael_work | includes* |
| 13:28.08 | [TK]D-Fender | SHOW us all of this... |
| 13:28.43 | michael_work | i prefer not to show my external IPs :) |
| 13:29.13 | Penguin | Oh boy. Another one of those. |
| 13:29.35 | Chainsaw | We prefer not to help redactionists. |
| 13:29.38 | Chainsaw | It makes our work impossible. |
| 13:29.39 | michael_work | in a sec |
| 13:29.56 | WIMPy | /whois michael_work |
| 13:31.17 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:37.28 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 13:39.53 | michael_work | i'm back :) |
| 13:46.02 | michael_work | preparing data |
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| 13:59.57 | *** mode/#asterisk [+o newtonr] by ChanServ |
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| 14:02.44 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 14:03.56 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 14:14.45 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 15:01.20 | michael_work | https://dl.dropboxusercontent.com/u/2336306/iax95.trace.pcap https://dl.dropboxusercontent.com/u/2336306/iax98.trace.pcap http://pastebin.com/CqmF9ACp |
| 15:01.48 | michael_work | WIMPy, [TK]D-Fender Penguin sorry for delay :( |
| 15:02.26 | [TK]D-Fender | You missed about half of what was requested and took and hour and a half to do it... |
| 15:02.33 | michael_work | so this is call from 192.168.1.95 to 192.168.1.98 and than opposite direction in same trace. traces on two servers |
| 15:02.44 | michael_work | i know |
| 15:02.47 | [TK]D-Fender | show us that * is even listening, that your peers are set up, IAX debug from CLI, etc |
| 15:02.53 | michael_work | i had other things i had to do - sorry |
| 15:03.32 | michael_work | they are not registered i'll paste configs |
| 15:04.51 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 15:06.46 | michael_work | [TK]D-Fender, http://pastebin.com/WcXgpaMp |
| 15:07.22 | michael_work | [TK]D-Fender, in CLI on calling side i see data, on reciving side nothing |
| 15:07.30 | michael_work | that's actually the problem |
| 15:07.41 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 15:07.41 | michael_work | it's not like it's rejecting or something it's like it's not listening at all |
| 15:12.16 | *** join/#asterisk CeBe (~CeBe@port-92-206-37-238.dynamic.qsc.de) |
| 15:15.00 | *** join/#asterisk CeBe (~CeBe@port-92-206-37-238.dynamic.qsc.de) |
| 15:16.08 | Penguin | rasterisk -x" module show " |grep iax? I think what you were looking for was asterisk -rx 'module show like iax" |
| 15:16.50 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 15:16.53 | [TK]D-Fender | no data = networking problem |
| 15:17.07 | [TK]D-Fender | Did you allow/forward the proper ports in? |
| 15:17.11 | Penguin | I'd still like to see that asterisk is listening on IAX2. |
| 15:17.22 | [TK]D-Fender | indeed |
| 15:17.27 | [TK]D-Fender | I did ask for that |
| 15:17.45 | Penguin | You know how that goes, though. |
| 15:19.15 | michael_work | Penguin, i tried modules like instead of module and wanted to make it as fast as i can |
| 15:19.30 | michael_work | [TK]D-Fender, but i do see on the side of the reciving server |
| 15:19.38 | michael_work | i applied tcpdukmp |
| 15:19.44 | Penguin | But WE DON'T see it. |
| 15:19.51 | michael_work | i sent you links |
| 15:20.01 | Penguin | I have learned that REGISTERs are good for troubleshooting routers/firewalls. |
| 15:20.02 | michael_work | <michael_work> https://dl.dropboxusercontent.com/u/2336306/iax95.trace.pcap https://dl.dropboxusercontent.com/u/2336306/iax98.trace.pcap |
| 15:20.12 | michael_work | no registers there |
| 15:20.42 | michael_work | it's invite only |
| 15:20.57 | michael_work | but .... if i do iax set debig on on reciver server i see noithing |
| 15:21.15 | michael_work | but i do see packets comming to server over tcpdump and ngrep |
| 15:21.17 | Penguin | I've had several networking problems recently, and the REGISTER and reply led me to the solution. |
| 15:21.17 | [TK]D-Fender | no packets = networking fail |
| 15:21.35 | michael_work | i know there are all kinds of people here but i'm not total noob, and yes i have problems of explaining myself :) |
| 15:21.38 | [TK]D-Fender | Where do we see * listening? |
| 15:22.02 | [TK]D-Fender | netstat -an|grep 4569 |
| 15:22.05 | michael_work | but i want host to be dynamic |
| 15:22.06 | Penguin | If the packets reach the host but don't show up in asterisk, that kind of points at asterisk being the bad link in the chain. |
| 15:22.19 | michael_work | the point of calls from different servers |
| 15:22.20 | Penguin | What does dynamic have to do with anything? |
| 15:22.34 | michael_work | and it works splendid on 1.8 |
| 15:22.40 | michael_work | same network config |
| 15:22.51 | Penguin | If you are talking about host=dynamic, that requires the peer to register. |
| 15:23.13 | [TK]D-Fender | And you'd still see IAx debug at * CLI if properly enabled either way |
| 15:23.26 | Penguin | Show that asterisk is listening on IAX2. |
| 15:23.35 | michael_work | i want to use same user/pass for different servers to connect to that one |
| 15:23.56 | michael_work | Penguin, netstats? |
| 15:24.04 | Penguin | netstat -lnpu |
| 15:24.21 | [TK]D-Fender | [11:22][TK]D-Fendernetstat -an|grep 4569 |
| 15:24.24 | Penguin | ^ |
| 15:25.00 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:25.06 | michael_work | netstat -an|grep 4569 |
| 15:25.06 | michael_work | udp 104192 0 0.0.0.0:4569 0.0.0.0:* |
| 15:25.12 | michael_work | <PROTECTED> |
| 15:25.13 | michael_work | udp 229376 0 0.0.0.0:4569 0.0.0.0:* |
| 15:25.17 | michael_work | 1st is 95 |
| 15:25.22 | michael_work | second is 95 |
| 15:25.25 | Penguin | If it isn't listening there, it will never show you anything coming to it. |
| 15:25.27 | michael_work | second is 98 |
| 15:25.30 | michael_work | sorry |
| 15:25.30 | Penguin | It has to LISTEN there. |
| 15:25.47 | michael_work | ^ |
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| 15:34.35 | [TK]D-Fender | those 2 pcaps are from the same server/direction |
| 15:34.45 | [TK]D-Fender | from 98 -> 95 |
| 15:34.52 | [TK]D-Fender | Where do we see 95's inbound? |
| 15:37.20 | michael_work | nope |
| 15:37.22 | michael_work | both |
| 15:37.27 | [TK]D-Fender | I jsut DL'd them |
| 15:37.37 | [TK]D-Fender | looks like you picked the wrong files |
| 15:37.42 | michael_work | i clade from 98 to 95 and than opposite |
| 15:37.44 | michael_work | same pcap |
| 15:37.51 | michael_work | 98 on 98 and 95 on 95 |
| 15:38.20 | [TK]D-Fender | I'm staring at them now... |
| 15:38.40 | [TK]D-Fender | the filename SAYS it's suppsed to be dirrect, but the source & dest columns are the same on both |
| 15:39.11 | Penguin | I would have logged on to 98 and done tshark host 95 and port 4569 |
| 15:39.24 | Penguin | I then would have logged on to 95 and done tshark host 98 and port 4569 |
| 15:41.32 | [TK]D-Fender | Something seems off here... |
| 15:45.12 | Penguin | example of how REGISTER is useful for network troubleshooting: http://pastebin.com/eC86mcsx |
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| 15:47.03 | Penguin | In my example, the network is in good shape in both directions. |
| 15:47.48 | Penguin | But it does require that asterisk is working properly. |
| 15:51.38 | Penguin | Is Heil Sound a popular name in the music/recording industry? |
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| 15:59.29 | michael_work | Penguin, [TK]D-Fender sorry for delay |
| 15:59.33 | michael_work | again :( |
| 15:59.41 | michael_work | i'm going off soon so i would not bother your time |
| 16:00.37 | michael_work | i just want to point out that the problem is that A calls B and i see packets of A on A and B while there is no answer on B side (not network not IAX debug not CLI debug) |
| 16:01.26 | michael_work | and if i call B to A it's same just swtched bettwen A and B in the above sentence |
| 16:01.32 | michael_work | it seems like asterisk ignores IAX at all |
| 16:01.34 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 16:02.02 | michael_work | i saw some rewlated bug but it said it was only on asterisk 11 -> 11.2 and was fixed |
| 16:02.07 | michael_work | on upcoming versions |
| 16:02.27 | michael_work | as well as there is no patch so as i don't have diff i can't check. |
| 16:03.10 | michael_work | if code is in |
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| 16:23.20 | npoulakos | hey, i having an issue using regular expressions in a if statment on ael. Im trying to change the caller id number if the number contains a-zA-Z. |
| 16:23.31 | npoulakos | right now my if statement look like, |
| 16:23.32 | npoulakos | if("${CALLERID(number)}" = [a-zA-Z]) |
| 16:24.07 | npoulakos | I tried a few other ways but im missing something |
| 16:27.03 | [TK]D-Fender | that isn't regex |
| 16:27.12 | [TK]D-Fender | that is a basic Asterisk expression |
| 16:27.44 | [TK]D-Fender | "core show function REGEX" |
| 16:37.05 | npoulakos | oh.. i think i see what i have to do. |
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| 16:49.04 | qakhan | hi i config sip trunk with provider but when i dial out there is message Got SIP response 484 "Address Incomplete" back from 10.200.7.157 |
| 16:49.11 | qakhan | and busy tone |
| 16:49.30 | [TK]D-Fender | They clearly don't like the number you are passing. |
| 16:50.52 | qakhan | ok so its mean we are not entering correct number |
| 16:51.14 | WIMPy | Possibly. |
| 16:51.26 | WIMPy | Entering and sending can be two different things. |
| 16:51.45 | WIMPy | But the 1st guess would be the format of the number. |
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| 16:52.16 | npoulakos | [TK]D-Fender: thanks for the help. i got it working. |
| 16:52.30 | [TK]D-Fender | npoulakos: Glad to hear. |
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| 17:12.52 | *** join/#asterisk workingcats (~workingca@212.122.48.77) |
| 17:17.35 | workingcats | hi, sorry for going a bit OT, i am looking for a VoIP client that can talk to asterisk, has a GUI *and* has the ability for some very basic "scripting". specifically, i need to be able to make calls (e.g. "ekiga -c SIPURL" does that part perfectly well) and also obtain basic call infos ("here are the last 5 calls, they started at these times, and were ended for these reasons")? |
| 17:18.14 | workingcats | even a simple phone status that merely says if it is currently in a call or not would do. would appreciate any pointers |
| 17:18.48 | workingcats | oh the client OS is linux. a derivative ubuntu 12.04 to be precise, but I'm happy to compile from source if necessary |
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| 17:25.26 | rrittgarn | look into isymphony for your GUI on top of a regular asterisk system... most cases it works "ok" we went on to write our own past that, but its a decent starting point... or flash operator panel, or i'm sure there are others. |
| 17:40.43 | workingcats | rrittgarn, cheers, having a look |
| 17:43.12 | *** join/#asterisk lorsungcu (~anonymous@67.138.198.66) |
| 17:45.33 | Penguin | I thought you were looking for a phone. |
| 17:46.38 | workingcats | yep, not quite what i need. found another approach though so i dont need that functionality from the phone anymore. would still be helpful though if someone has another tip |
| 17:50.04 | workingcats | as in, i don't *need* it anymore due to features of the system i will be writing to, but it would still be very useful. i'll stick around just in case someone can help me, or asks a question i can answer ;) |
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| 17:55.16 | Penguin | workingcats: I like twinkle. I don't know about the scripting part of your request, but I'm pretty sure it does the other stuff. |
| 17:57.09 | workingcats | Penguin, doesnt seem to be maintained anymore, at least going by the front page at twinklephone.com |
| 17:57.27 | Penguin | I don't recall that being part of your requirements. |
| 17:57.31 | workingcats | hehe |
| 17:57.35 | workingcats | no, it wasn't |
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| 19:58.23 | jmmills | channel debug shows dtmf, have a feature in my applicationmap that calls playback, set the DYNAMIC_FEATURES variable to the name of that test feature, setup a test call, hit the dtmf sequence and nothing, not even on console |
| 19:58.31 | jmmills | There has to be something obvious I'm missing? |
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| 20:38.33 | newtonr | jmmills, pastebin the output of "logger show channels" and "core show settings" |
| 20:40.20 | jmmills | @newtonr https://gist.github.com/jmmills/9536540 |
| 20:41.38 | newtonr | jmmills, well, you don't have DTMF, VERBOSE or DEBUG going to the console, so you shouldn't expect to see anything going there |
| 20:42.01 | jmmills | I got DTMF when I debuged the channel |
| 20:42.21 | newtonr | what kind of channel? Is this SIP? |
| 20:42.26 | jmmills | yup |
| 20:42.53 | jmmills | I just happened to dump you settings after I turned off debugging |
| 20:43.25 | newtonr | jmmills, Okay, so you know there is DTMF. Is the problem that your dynamic feature is not triggered after the DTMF sequence is entered? |
| 20:43.38 | jmmills | yes |
| 20:43.53 | jmmills | and I did a dialplan set chanvar DYNAMIC_FEATURES testfeature |
| 20:43.58 | jmmills | where my feature is testfeature |
| 20:44.23 | jmmills | ynamic Feature Default Current |
| 20:44.23 | jmmills | --------------- ------- ------- |
| 20:44.24 | jmmills | testfeature no def #3 |
| 20:44.28 | jmmills | so it exists |
| 20:47.07 | newtonr | set debug and verbose to 5 in asterisk.conf, restart Asterisk, try again, and pastebin the log *with SIP debug*, DEBUG and DTMF showing in it |
| 20:48.22 | jmmills | okay, gimme a min |
| 20:48.25 | jmmills | stupid RTS |
| 20:48.34 | jmmills | I am not a fan of it |
| 20:48.53 | jmmills | at least for things out out side of dialplans |
| 20:49.10 | jmmills | but then again agi makes database loading a dialplan kind of moot |
| 20:55.08 | jmmills | grrg, going ommit some of this junk |
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| 21:18.30 | jmmills | newtonr: had to trim it down due to pastebin size limits - http://pastebin.com/KFZfLe4x |
| 21:18.40 | jmmills | I also omitted ip addresses |
| 21:18.48 | jmmills | err redacted? |
| 21:24.29 | newtonr | jmmills, I'll take a look |
| 21:24.38 | jmmills | I really don't see anything there |
| 21:25.04 | jmmills | I'm attempting to setup a test instance that is pretty default to see if I can even get it to work |
| 21:25.21 | jmmills | that will tell me if it has something to do with this kunckleheaded setup - or something to do with the version of asterisk |
| 21:26.12 | newtonr | yeah asterisk is seeing the DTMF, I just don't see a feature being triggered |
| 21:27.17 | newtonr | jmmills, simplifying your setup is a good next step |
| 21:27.44 | jmmills | yeah the one I logged to you is the "test pbx" in a 7 pbx federation |
| 21:27.55 | jmmills | and by test it just isn't in the SRV failover |
| 21:29.29 | jmmills | okay, well first attempt at dynamic feature is a failure |
| 21:29.33 | jmmills | behavior is consistent |
| 21:29.42 | jmmills | which indicates a misconfiguration of some kind |
| 21:30.30 | Katty | i'm going to bet on a drug interaction issue. |
| 21:30.50 | jmmills | Are you saying I should get high and try and fix this? |
| 21:31.01 | Katty | mmm, no. i'd try .25mg of xanax |
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| 21:33.22 | Katty | hi tony |
| 21:33.44 | Katty | we were just debating how much xanax to apply to jmmills's problem. |
| 21:34.09 | jmmills | plural |
| 21:34.13 | jmmills | I have more than one problem :) |
| 21:35.37 | Katty | bind to All The GABA receptors! |
| 21:36.27 | Katty | actually that'd be a terrible idea. |
| 21:36.48 | jmmills | I know next to nothing about neurology |
| 21:37.28 | Katty | me either. |
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| 21:59.30 | tuxd00d | Is there a good page that explains sip types, "peer", "user" and "friend"? |
| 22:00.36 | *** join/#asterisk talnti (~talntid@216.229.186.226) |
| 22:00.45 | talnti | I have in queues.conf, strategy=roundrobin |
| 22:00.52 | jmmills | tuxd00d: No. |
| 22:00.57 | jmmills | I still get confused by them |
| 22:01.09 | talnti | but: operator has 0 calls (max unlimited) in 'rrmemory' strategy (13s holdtime, 36s talktime), W:0, C:111, A:19, SL:0.0% within 0s |
| 22:02.00 | tuxd00d | jmmills: Thanks :) Would you take a guess what my outgoing and incoming trunks should be set to? |
| 22:02.42 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 22:02.53 | jmmills | It depends on your callflow |
| 22:02.59 | tuxd00d | I don't know if this is still (or ever was) accurate. http://www.voip-info.org/wiki/view/Asterisk+sip+type |
| 22:03.10 | jmmills | generally you have to build a user and a peer for bidirectional calls |
| 22:03.27 | jmmills | friend is the easy way, but it's also easy to be very insecure with friend |
| 22:03.42 | tuxd00d | jmmills: they are unidirectional. My only bidirectional would be my extensions. |
| 22:04.17 | jmmills | So you are not recieving INVITES from your trunks, just sending them? |
| 22:04.18 | tuxd00d | I set my incoming truck to |
| 22:04.23 | *** join/#asterisk aness (~aness@cm-84.215.76.18.getinternet.no) |
| 22:04.45 | tuxd00d | I set my incoming truck to 'user' |
| 22:04.53 | tuxd00d | and it works fine |
| 22:05.09 | elguero | jmmills: Do you set DYNAMIC_FEATURES or _DYNAMIC_FEATURES? |
| 22:05.17 | jmmills | elguero: I've tried both |
| 22:05.18 | tuxd00d | Wait, sorry, I mean I set my outgoing to peer, and it works fine |
| 22:05.32 | jmmills | well I didn't on the isolated |
| 22:05.37 | jmmills | is it _ or two _ |
| 22:05.37 | jmmills | __ |
| 22:05.57 | elguero | just one |
| 22:06.14 | jmmills | tuxd00d: well there you go, but you will probably throw a 401 to that trunk if tries to send a call |
| 22:06.23 | jmmills | elguero: I'll try again again |
| 22:06.31 | tuxd00d | My incoming, set at user, doesn't work, chan_sip.c:25576 handle_request_invite: Failed to authenticate device "+XXXXXXXXXX" <sip:XXXXXXXXXX@XX.XX.XX.XX> |
| 22:07.38 | jmmills | elguero: eureeka! |
| 22:07.47 | jmmills | mother of undocumented |
| 22:07.51 | jmmills | the wiki I think had two |
| 22:07.58 | elguero | what wiki? |
| 22:08.39 | jmmills | voipinfo |
| 22:09.19 | elguero | ah... stop using voipinfo :) |
| 22:09.47 | elguero | wiki.asterisk.org |
| 22:10.12 | elguero | https://wiki.asterisk.org/wiki/display/AST/Variable+Inheritance+Basics |
| 22:15.59 | tuxd00d | jmmills: Just FYI, part on sip type matching http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeviceConfig_id216341.html#DeviceConfig_id291081 |
| 22:28.27 | *** join/#asterisk matrix1233 (~matrix123@41.228.40.3) |
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| 22:30.53 | matrix1233 | hello, i have sip extension from 2000 to 2100 and i wanna ti do a spygroup from 2001 to 2010 like for example when i call 556 i will spy on random 2001 to 2010 and when i call 557 i will spy 2011 to 2020 |
| 22:31.00 | matrix1233 | how can i do that |
| 22:32.27 | rrittgarn | core show application chanspy |
| 22:33.10 | matrix1233 | rrittgarn: i have read the doc but can help to understand more |
| 22:33.42 | matrix1233 | rrittgarn: i know that i need to use SPYGROUP |
| 22:35.06 | file | looks around |
| 22:36.00 | rrittgarn | well, i've not used it... but reading the documentation on it, it looks like the spygroup variable is for limiting who can actually do the spying. So when you start a channel and set the spygroup variable to a group name with something like set(SPYGROUP=Group1) Doesn't look like it does anything random |
| 22:38.06 | rrittgarn | so if you had a bunch of calls coming through you could set that value, then do a chan spy into that group and cycle through them |
| 22:38.17 | rrittgarn | you'd just have to set the variable before calling chanspy |
| 22:38.44 | rrittgarn | correction: specify the g() option |
| 22:39.06 | matrix1233 | rrittgarn: ok but how can say that the Group1 have from 2001 to 2010 |
| 22:39.41 | rrittgarn | set it up so when calls go to those extensions they get flagged in that group |
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| 22:42.48 | matrix1233 | rrittgarn: thanks |
| 22:43.29 | rrittgarn | np |
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| 22:58.17 | tuxd00d | jmmills: http://burner.com/asterisk-primer/configuring-sip/ |
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| 23:44.07 | jmmills | Well, now I'm just confused |
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| 23:45.57 | jmmills | def getting dtmf, running the same features.conf as my successful test |
| 23:46.07 | jmmills | checks for bugs in current asterisk version |
| 23:46.47 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 23:48.35 | m0sphere | my customer is trying to send a fax through DISA. according to my logs they enter a fax number, then i see FAX CNG detected but no fax extension, it appends "f" to the dialed number, and my server responds with cannot complete as dialed. how do i fix this? |
| 23:49.33 | WIMPy | Interesting. |
| 23:49.43 | WIMPy | But why are you using DISA? |
| 23:50.34 | m0sphere | becase I allow my customers to use DISA to dial out from my PBX |
| 23:50.54 | m0sphere | it's kinda like a calling card, but set up a little differently |
| 23:51.00 | WIMPy | And why don't they just dial? |
| 23:51.06 | m0sphere | they dial in, then dial out |
| 23:51.26 | leifmadsen | this doesn't go directly to what you're trying to solve, but I honestly prefer the Read() application for this kind of input |
| 23:51.29 | leifmadsen | you have a lot more control there |
| 23:51.35 | leifmadsen | and you can still provide indications through the 'i' flag |
| 23:51.49 | *** join/#asterisk jmls (~julian@77.107.171.82) |
| 23:51.52 | WIMPy | WaitExten would be an alternative. |
| 23:52.09 | WIMPy | But Read isn't usefull for dialling. |
| 23:52.22 | leifmadsen | sure it is |
| 23:52.25 | leifmadsen | I use it all the time for that |
| 23:52.40 | WIMPy | If you like timeouts. |
| 23:52.47 | leifmadsen | just set no timeout then |
| 23:52.50 | WIMPy | And customers who moan about that. |
| 23:53.07 | WIMPy | And how do I know when the entered number is complete? |
| 23:53.17 | leifmadsen | I'm not in Europe |
| 23:53.21 | leifmadsen | I have a fixed input |
| 23:53.22 | leifmadsen | so I know |
| 23:53.38 | WIMPy | And you don't want to call there, either, I guess. |
| 23:53.47 | leifmadsen | in my case, that is not an issue |
| 23:54.06 | leifmadsen | you're applying a logical problem to an application |
| 23:54.33 | leifmadsen | there are multiple ways to solve problems, and in many cases, Read() is a perfectly viable alternative to DISA() with additional control |
| 23:55.25 | leifmadsen | m0sphere: honestly your issue sounds like a bit of a bug with DISA(). Maybe provide from console output to see if that is truly the case, or if there might be a work around |
| 23:56.22 | m0sphere | is it a bug with disa, or could the F be appended by the users fax machine, to indicate fax in some way? |
| 23:56.32 | leifmadsen | m0sphere: hard to say without a console output |
| 23:56.41 | leifmadsen | there are only DTMF A-D afaik :) |
| 23:56.56 | leifmadsen | and since you're using DISA(), it would have to be DTMF input |
| 23:57.46 | WIMPy | Maybe you should just switch off fax detection. |
| 23:58.52 | m0sphere | i had fax detection off, thinking that was the issue i turned it on, but got the same result |
| 23:59.10 | m0sphere | and unforunately I have no fax machine to test with |
| 23:59.39 | m0sphere | i wish people didn't still use fax machines |
| 23:59.48 | WIMPy | Oh, I like that. When you try to evaluate an issue you hit multiple oterhs. |
