| 00:01.26 | ChannelZ | I'd have to guess on a business day. As in tomorrow. |
| 00:01.29 | tyman | Wasn't expecting anything official here…sent email to orders@digium. |
| 00:02.43 | tyman | ChannelZ: bummer…seems easily automated. |
| 00:03.10 | ChannelZ | You'd think |
| 00:05.11 | ChannelZ | I'm just guessing though. To be honest I can't remember, like when I got an FFA license or the old Skype license, if it was totally automated or I had to wait. |
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| 00:45.12 | GerNewb | someone online? |
| 00:45.22 | GerNewb | and can help me |
| 00:46.12 | [TK]D-Fender | If you tell us the problem ... maybe |
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| 02:59.13 | volga629 | How I can check dialled remote extension, if extension starts from specific number |
| 03:06.15 | ChannelZ | huh? |
| 03:09.43 | volga629 | example 724XXX how I can check extension which was dialled, I mean from which number is starts, based on this put call through different trunk |
| 03:11.37 | ChannelZ | well ${EXTEN} is the extension dialed, you can look at it like any other time. |
| 03:13.30 | volga629 | exten => s,n,GotoIf(${IF($[${EXTEN} = "^7"])?dest-amst-remote) some thing like this ? |
| 03:13.55 | ChannelZ | well that's a regex |
| 03:14.08 | ChannelZ | if you only wanted to see if it started with a 7, ${EXTEN:0:1} would give you the first digit |
| 03:14.13 | volga629 | yes begin if |
| 03:14.16 | volga629 | with |
| 03:15.07 | volga629 | it like parameter expansion in bash ? |
| 03:16.21 | volga629 | yes, remote branches have call prefix like 7 and 9, based on this I can route call easy |
| 03:16.24 | ChannelZ | it's character selection |
| 03:16.39 | volga629 | yes let me try |
| 03:16.51 | ChannelZ | ${VAR:START:LEN} |
| 03:17.08 | volga629 | Or I see |
| 03:17.25 | ChannelZ | ${EXTEN:1} cuts off the first digit, ${EXTEN:-1} gives the last digit, ${EXTEN:0:1} gives the first, etc. |
| 03:18.27 | volga629 | ${EXTEN:1} this will valid extension 72449 without 7. |
| 03:22.41 | ChannelZ | If the extension is 72449 ${EXTEN:1} would be 2449 |
| 03:22.47 | ChannelZ | yes |
| 03:24.28 | volga629 | ok let me try to apply |
| 03:25.55 | WIMPy | Looks like what the patterns would do for you. |
| 03:33.05 | ChannelZ | there is a regex operator but for this, not worth the bother |
| 03:34.09 | WIMPy | But if I want the first digit, I'd probably use multiple extensions. |
| 03:34.52 | WIMPy | Unless I do some clever stuff, generating global variable names. |
| 03:35.19 | volga629 | yes, all my task is try send message with MessageSend trough trunk to another box |
| 03:37.54 | volga629 | as fa I understand MessageSend() only use URI and I can't use SIP/trunk/extension |
| 03:37.59 | volga629 | far |
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| 03:46.08 | volga629 | exten => s,n,Set(TO=${OUT_2}/${EXTEN:1}) <-- Incorrect, How I cant put this correctly / |
| 03:46.11 | volga629 | ? |
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| 04:02.07 | ChannelZ | well, the extension in your example is 's' so cutting off the first digit of s doesn't make a lot of sense. |
| 04:02.35 | ChannelZ | s minus s is nothing |
| 04:02.41 | volga629 | ${EXTEN}: The current extension - cannot be modified with the set command- just use the GoTo to change the EXTEN variable! |
| 04:02.47 | volga629 | Oooooo |
| 04:04.37 | volga629 | yes, you right just looking how to put right way |
| 04:05.38 | ChannelZ | well put it on an extension that has more than 1 digit, presumably.. like exten => _7XXXX,1,NoOp(${EXTEN:1} is something dialed minus the 7) |
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| 04:08.07 | volga629 | that the close what I putted |
| 04:08.40 | volga629 | https://fpaste.networklab.ca/bEqc/ |
| 04:11.36 | volga629 | only last step I don't know how to set for MessageSend uri |
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| 04:47.34 | devdvd | Can I use templates in sip.conf for the phones and still use realtime? What I want to do is define a set of parameters (a template) in sip.conf but have the actual phone definitions (user,password,extension) read in from the database |
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| 05:23.19 | [TK]D-Fender | devdvd: nope |
| 05:24.22 | [TK]D-Fender | volga629: What is there ti "set"? Just pass it what you want to pass it |
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| 05:25.52 | devdvd | tk, thanks |
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| 05:55.46 | bulkorok | good morning |
| 05:59.22 | Maliuta | no, it's late afternoon |
| 05:59.24 | Maliuta | :P |
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| 07:03.39 | tyman | is there a channel best suited to ask questions about DPMA? |
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| 08:17.57 | ThothCastel | How would one go about to add a recovery delivery to deliver incoming calls to a mobile phone number in case the isdn line goes down? |
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| 09:29.08 | danfromuk | Hi, I just had to change a network card in our asterisk box and now the g729 licenses say 'incorrect host id'. How can I fixing that quickly? |
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| 10:07.24 | pirtin | Hello. |
| 10:10.02 | pirtin | I've got a question which seems dumb to me, but I can't find the answer. When a message is displayed in the CLI, for exemple 'WARNING[1188][C-00000c4f]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)' |
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| 10:13.05 | pirtin | How could I have more informations on this message ? What is the first number inside square brackets reffering to ? |
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| 10:25.00 | wdoekes | pirtin: thread/process id |
| 10:28.10 | Faustov | does anyone know if meetme/confbridge can be configured to work similarly to webex? |
| 10:28.27 | wdoekes | (fyi: gettid() returns the caller's thread ID (TID)) |
| 10:28.43 | Faustov | ie booking returns a dynamic conference number with a code that gets auto-distributed via email |
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| 10:33.36 | wdoekes | Faustov: confbridge is dumb enough to do that. add custom dialplan code to suit your needs |
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| 10:37.57 | Mads | Hello. Can someone explain to me what the following line does? I'm not sure i understand the syntax --> Set(NUMBER=${IF($[${LEN(${EXTEN:4})}>8]?${EXTEN:4:-2}:${EXTEN:4})}); |
| 10:41.29 | Faustov | wdoekes: thanks, i thought so (that I'd have to move from meetme to confbridge) |
| 10:42.16 | pirtin | wdoekes, thanks. I try raising verbose to see if it could give me what i want. |
| 10:42.16 | Gugge | Mads Set(NUMBER=) sets the variable NUMBER |
| 10:44.12 | Gugge | Mads: ${IF($[${LEN(${EXTEN:4})}>8]?${EXTEN:4:-2}:${EXTEN:4})} - if ${LEN(${EXTEN:4})}>8 then ${EXTEN:4:-2} else ${EXTEN:4} |
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| 10:57.06 | Mads | Gugge i can see what it does there but i dont know what EXTEN:4 is can you maybe explain that to me? |
| 11:00.45 | Gugge | VAR:offset:length |
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| 12:32.13 | Vico100 | hello |
| 12:32.27 | Vico100 | can asterisk dial out to a voip (sip) provider? |
| 12:32.56 | Greenlight | Vico100: Yes. |
| 12:33.51 | Vico100 | i am beginning asterisk, my cisco phone can call another cisco phone connected to the network via asterisk |
| 12:34.12 | Vico100 | but how do i register asterisks to the voip provider (via the internet) |
| 12:34.18 | Greenlight | ~book |
| 12:34.18 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
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| 12:34.26 | Vico100 | book to hard ^^ |
| 12:34.27 | Vico100 | okay |
| 12:34.40 | Vico100 | 800+ pages |
| 12:34.41 | Vico100 | hehe |
| 12:34.49 | Greenlight | You don't need to read it all. |
| 12:35.09 | Vico100 | ok |
| 12:39.39 | [TK]D-Fender | Vico100: There are hundreds of guides for dozens of different providers out there.... a SIP peer entry is a SIP peer entry. a registration is a registration. It's all the same. |
| 12:40.13 | [TK]D-Fender | Vico100: What your provider requires in addition to any other is up to your provider. See if they offer their own sample. If not, stick with the basics to start |
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| 12:41.32 | Vico100 | i know how to register to the sip with the cisco phone, or simply an ata, i have the url, port, username password, ect. I just don't kow in watch config file to put it in. |
| 12:41.52 | Vico100 | *which |
| 12:42.15 | Greenlight | sip.conf, as you have with the phones. |
| 12:42.38 | Greenlight | To asterisk it's just an endpoint, same as your phones |
| 12:43.10 | Vico100 | okay thankyou. |
| 12:43.16 | Vico100 | ah thankyou |
| 12:43.22 | Vico100 | i did not think of it that way |
| 12:43.36 | Vico100 | wow its so much easyier now :) |
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| 13:03.54 | dose | hey guys |
| 13:04.10 | dose | dumb issue that I couldn't figure out |
| 13:04.25 | dose | how to hide CLI for outbound calls from analog phone :) |
| 13:05.07 | [TK]D-Fender | dose: "analog phone" has nothing to do with your outbound. |
| 13:05.25 | [TK]D-Fender | dose: For the aprt of your dialplan that actually does the calling out... that is where you do it. |
| 13:05.31 | [TK]D-Fender | part* |
| 13:06.39 | [TK]D-Fender | dose: As for the actual hiding.. that depends what youare calling OUT over |
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| 13:10.00 | dose | [TK]D-Fender: Analog phone is using same dialplan than SIP phone for outbound calls over PRI link, when it's a SIP phone I check if property of phone is using prohib based on realtime settings of the extension but for analog phone I don't know how to do it without having a separate dialplan. I have tried to configure |
| 13:10.28 | dose | hidecallerid=yes restrictcid=yes on the dahdi-channels on the analog extension itself but doesn't works |
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| 13:11.55 | [TK]D-Fender | dose: You're either going to have to make some checks in the dialplan to adapt to the device that is calling, or make separate dialplan altogether |
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| 13:12.40 | dose | how boring there is no easy way to do it on analog phone itself than manipulating all dialplan to handle one analog phone |
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| 13:16.31 | [TK]D-Fender | Unless your dialplan is a mess, accounting for one phone should be easy |
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| 13:18.58 | dose | no it's not a mess |
| 13:19.07 | dose | ok thanks for the indication |
| 13:19.14 | dose | I'll handle it into dialplan |
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| 13:20.40 | puzzled | hi |
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| 13:29.27 | igcewieling | Ugh. Satan Herself is asking customer support. |
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| 14:40.52 | Ice_Strike | "-- Got 0ms silence< 2500ms required" this getting flooded in asterisk CLI |
| 14:40.55 | Ice_Strike | where is that come from? |
| 14:42.21 | igcewieling | Ice_Strike: the client |
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| 15:10.29 | Greenlight | Ice_Strike: AMD |
| 15:10.34 | Greenlight | Or WaitForSilence |
| 15:10.58 | Greenlight | Or similar dialplan application |
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| 16:12.32 | ABerrios | Does Record() record dtmf tones? I see the x option ignores dtmf as a terminator but that's besides the point. |
| 16:13.44 | WIMPy | Depends on how you receive DTMF. It will record all audio. |
| 16:14.09 | ABerrios | guess that answers my Q, thanks WIMPy |
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| 16:41.02 | *** join/#asterisk przerull (~philip@philzerull.com) |
| 16:41.42 | przerull | hello, so when I try using the CUT function in ael, I get the following notices in my cli |
| 16:41.45 | przerull | <PROTECTED> |
| 16:41.51 | przerull | ast_expr2.y:782 compose_func_args: argbuf uses 11 bytes; |
| 16:41.59 | przerull | however the function works just fine |
| 16:42.05 | przerull | should I be concerned about this |
| 16:42.28 | Greenlight | You have debug enabled or something ? |
| 16:43.03 | przerull | I don't beleive so. no. my verbose is 0 and I shouldn't have any debug's turned on |
| 16:43.27 | Greenlight | Odd that you're getting that with verbose 0 |
| 16:43.39 | Greenlight | And without debug enabled |
| 16:44.17 | przerull | it's a NOTICE message |
| 16:45.00 | Greenlight | Hmm which version are you running ? |
| 16:45.07 | igcewieling | przerull: Then it should be harmless |
| 16:45.32 | Greenlight | Indeed, it's certainly harmless, but still odd that it's showing |
| 16:46.02 | przerull | no, just looked my logger.conf has console => notice,warning,error |
| 16:46.14 | przerull | i'll probalby remove the notice from it |
| 16:47.06 | przerull | the bigger issue I'm experiencing is that I get a ton of garbage over agi |
| 16:47.09 | przerull | from asterisk |
| 16:47.34 | Greenlight | Define "garbage"... |
| 16:47.57 | Greenlight | It's supposed to send a load of information when it first connects |
| 16:48.14 | przerull | here's an example of what I'm receiving |
| 16:48.22 | przerull | this was after the initial connect |
| 16:48.41 | przerull | my agi application wanted to get the channel variable HANGUPCAUSE |
| 16:48.44 | przerull | <PROTECTED> |
| 16:49.01 | *** join/#asterisk orthoguy (62bd7a33@gateway/web/freenode/ip.98.189.122.51) |
| 16:49.21 | Greenlight | Sure you're not doing something wrong in your app ? |
| 16:49.39 | Greenlight | That looks like it's outputting the hex |
| 16:51.11 | Greenlight | Or like theres a mismatch between binary and ascii |
| 16:51.14 | przerull | my app is pretty simple. I just pull bytes of the network until my message ends with two consecutive linefeeds |
| 16:51.59 | przerull | what's the encoding asterisk uses for AGI? |
| 16:52.50 | *** join/#asterisk cmendes0101 (~cmendes01@72.1.46.254) |
| 16:53.04 | Greenlight | I don't recall off hand |
| 16:53.31 | Greenlight | There's ready-made libraries that make things easier though |
| 16:53.48 | igcewieling | przerull: what AGI library are you using? |
| 16:54.04 | Greenlight | I get the impression he's doing it from scratch without lib |
| 16:54.11 | igcewieling | Greenlight: then he is a fool. |
| 16:54.44 | Greenlight | I thought you code was you wishing someone good luck... |
| 16:55.51 | Greenlight | In fairness I'm pretty sure I've got a "raw" AGI script around that I used somewhere as well |
| 16:56.07 | przerull | ha guilty...somewhat |
| 16:57.52 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-osjtxfulhlrvejtx) |
| 16:58.29 | przerull | however my code used to work great. was working great for months, then I made some changes to the dialplan on asterisk's end and now I get all kinds of strange things |
| 16:58.50 | igcewieling | przerull: you are strictly following the AGI protocol? |
| 16:59.21 | igcewieling | such as reading stdin when you script starts, to accept all the data Asterisk sends to the AGI script on startup? |
| 16:59.23 | przerull | from what I gathered from the documentation. I'm posting my method to a gist |
| 17:00.13 | igcewieling | will assume a gist is a form of a pastebin |
| 17:00.36 | przerull | yeah |
| 17:00.38 | przerull | https://gist.github.com/przerull/6105813 |
| 17:01.11 | *** join/#asterisk vlad_starkov (~vlad_star@176.74.9.254) |
| 17:01.14 | przerull | this is run by providing a socket object and a termination string. I send double line feeds |
| 17:01.34 | Greenlight | Soooo |
| 17:01.47 | Greenlight | If there wasn't a double like feed, what's to stop keeping reading into memeory ? |
| 17:01.51 | Greenlight | *memory |
| 17:01.58 | Greenlight | As in, unallocated, random stuff |
| 17:02.09 | Greenlight | Since, there's no length specified |
| 17:02.14 | *** join/#asterisk |Syd| (~Syd@unaffiliated/syd/x-8369111) |
| 17:02.25 | przerull | why wouldn't there be a double line feed? doesn't asterisk follow it's own protocol? |
| 17:02.49 | przerull | the agi protocol doesn't specify any length |
| 17:03.01 | Greenlight | Umm |
| 17:03.12 | Greenlight | new_chunk = some_socket.recv(8192) |
| 17:03.18 | igcewieling | I don't see the emptying of stdin |
| 17:03.23 | przerull | this isn't using stdin |
| 17:03.24 | Greenlight | What makes you think that ENDS with the double line feed |
| 17:03.28 | przerull | it's a tcp socket |
| 17:03.29 | Greenlight | ^^^ |
| 17:03.52 | Greenlight | In fact, I know that sometimes a packet can contain multiple messages |
| 17:04.12 | Greenlight | You really need to parse it byte by byte if you're doing it that way |
| 17:04.32 | Greenlight | If you run tcpdump, it'll likely help you diagnose what's going on |
| 17:04.46 | igcewieling | It isn't a socket, it is standard in. |
| 17:05.13 | Greenlight | Oh, this isn't "remote" AGI ? |
| 17:05.21 | Greenlight | I just presumed as it mentioned a socket |
| 17:05.41 | przerull | this is "remote" AGI |
| 17:05.44 | igcewieling | Greenlight: Even fastagi uses stdin for your AGI. The FastAGI wrapper handles all the socket stuff. |
| 17:05.46 | przerull | it call it over port 771 |
| 17:05.54 | przerull | 7771 |
| 17:05.59 | igcewieling | przerull: there is no such thing as a "remote AGI". Do you mean FAST AGI? |
| 17:06.04 | przerull | yes |
| 17:06.06 | Greenlight | Yea, FastAGI |
| 17:06.10 | igcewieling | show us the dialplan line? |
| 17:06.16 | Greenlight | That's not how I understood it, but perhaps I'm mistaken |
| 17:06.17 | igcewieling | then fscking call it what is. |
| 17:06.47 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 17:07.02 | przerull | agi(${CONST_AGI_PATH}/Register?callid=${callid}&channel_type=${channel_type}); |
| 17:07.11 | igcewieling | http://stackoverflow.com/questions/14806125/programming-fastagi-for-asterisk-using-python http://www.voip-info.org/wiki/view/Asterisk+FastAGI |
| 17:07.26 | przerull | CONST_AGI_PATH=agi://127.0.0.1:7771; |
| 17:07.26 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 17:07.44 | igcewieling | przerull: that would be the problem. You need an exten => line with a FastAGI (or just AGI) app. |
| 17:08.19 | przerull | this is ael |
| 17:08.33 | igcewieling | you keep talking, but your words make no sense |
| 17:08.35 | przerull | sorry, i prefer ael over the raw dialplan launguage |
| 17:08.35 | [TK]D-Fender | I thouch that when you specify a URI for AGI that is inherently called FastAGI |
| 17:08.47 | Greenlight | It is |
| 17:08.53 | igcewieling | [TK]D-Fender: only in 1.6+ (or maybe 1.4+) |
| 17:09.02 | Greenlight | Clearly it's getting that far at least... *something* is connecting to his app |
| 17:09.10 | Greenlight | Now, as I advised, grab a tcpdump |
| 17:09.19 | Greenlight | It looks like a simple reading error |
| 17:09.22 | igcewieling | przerull: when you paste the dialplan or AEL call to FastAGI/AGI then I'll continue helping, otherwise you are on your own. |
| 17:09.37 | Greenlight | [06:07pm] <przerull> agi(${CONST_AGI_PATH}/Register?callid=${callid}&channel_type=${channel_type}); |
| 17:10.47 | przerull | igcewieling: was that line of code not sufficient. I don't see how it's applicable to this issue? |
| 17:10.56 | Greenlight | I've already highlighed at least one bug in the way you're reading from the socket |
| 17:11.16 | igcewieling | I missed that. |
| 17:12.22 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-fzzvynxbxnvpveqg) |
| 17:12.34 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 17:12.39 | igcewieling | przerull: good! I suppose I should verify the fastAGI options did not change between Asterisk versions, but I'm sure you've read all that. |
| 17:12.46 | Greenlight | przerull: Imagine your take the following *from* the socket into new_chunk. "messagecontent\r\n\r\nmessage2content\r\nmessage3contenet" |
| 17:12.57 | *** join/#asterisk Minotaur01 (~minotaur0@199.119.232.225) |
| 17:13.08 | Greenlight | Now, that will get parsed incorrectly by your app... |
| 17:13.21 | Greenlight | So, fix that and then if you still have issues, grab a tcpdump |
| 17:15.24 | *** join/#asterisk transfinite (~aleph_0@unaffiliated/transfinite) |
| 17:15.39 | igcewieling | Greenlight: I think tcpdump will be enough to show him Asterisk is sending plain ASCII |
| 17:16.04 | Greenlight | Indeed |
| 17:17.46 | przerull | thanks. i'm pulling a tcpdump now. it might take a bit for me to get another goofy message |
| 17:18.14 | Greenlight | And you fixed that bug ? |
| 17:18.56 | przerull | double line feeds do end the message correct? |
| 17:19.10 | Greenlight | ... |
| 17:19.21 | Greenlight | Depends on your defenition of message |
| 17:19.22 | przerull | even though multiple messages can be sent on the same socket |
| 17:19.54 | Greenlight | Your code assumes that you'll never get sent more than one 1 message at once. This is an incorrect assumption. |
| 17:20.33 | Greenlight | Although it's an assumption that will hold true a *lot* of the time. |
| 17:23.42 | przerull | that is a good point. i'll have to consider a modification for that. I have been careful in my app to send a request and wait for a finished result before pulling more off the socket, however that's really not a safe way to do it |
| 17:23.59 | przerull | still waiting on tcpdump |
| 17:24.09 | przerull | (my prod box didn't even have it installed) |
| 17:24.36 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 17:25.09 | Greenlight | You want to take it a byte/char at a time, and check for occurance of your termination string, in which case you also need to keep whatever is "left" from your socket read, for the next message. |
| 17:25.43 | Greenlight | Or alternatively, as igcewieling suggested, use one of the libraries/wrappers, which abstract you from this. |
| 17:32.56 | WIMPy | Don't read one char at a time. That will be horribly slow. |
| 17:33.51 | WIMPy | Read as much as you can, process as far as the message goes and keep what's left over for the next round. |
| 17:34.08 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 17:34.25 | Greenlight | "process as far as the message goes" <-- And that's not going to look through the string one char at a time anyway? |
| 17:34.42 | WIMPy | That's a different matter. |
| 17:35.06 | Greenlight | I don't see how. We're talking about string manipulation. |
| 17:35.08 | WIMPy | No system calls involved there. |
| 17:35.20 | Greenlight | Oh,not from the socket not, from the buffer |
| 17:35.24 | Greenlight | *no |
| 17:35.32 | WIMPy | I was talking about reading input. |
| 17:35.39 | WIMPy | Ok. |
| 17:35.45 | danfromuk | Hi, a clients have intermittent problems where they can't hear anything and no one can hear them. When it occurs, the CLI displays this. http://pastebin.com/qGZ5jNqH |
| 17:35.46 | Greenlight | He's reading from a socket |
| 17:35.49 | WIMPy | Then we're on the same route. |
| 17:35.52 | danfromuk | Any ideas what the cause might be? |
| 17:35.54 | Greenlight | Yup :) |
| 17:36.19 | Greenlight | danfromuk: Seems your endpoints are trying to use g729 |
| 17:36.31 | danfromuk | They are. Thats ok. We have licenses. |
| 17:36.43 | Greenlight | Not according to that CLI |
| 17:36.44 | WIMPy | Enough? |
| 17:37.00 | Greenlight | Remember you'll need more than you think if you're MixMonitoring etc |
| 17:37.43 | danfromuk | We have 7 licenses. This client is the only client to use g729, and they only have 2 channels so should be plenty. |
| 17:38.20 | Greenlight | It seems it's changed mid-call somehow though |
| 17:38.37 | danfromuk | just scrolling back through the logs to the start of the call to see. |
| 17:38.47 | Greenlight | BUt I'm not really sure how that error could be generated, sorry |
| 17:39.06 | Greenlight | Dropping incompatible voice frame on Local/4367682553133@outgoing_ratelist0-000004d5;1 of format alaw since our native format has changed to (g729) |
| 17:39.17 | danfromuk | Ok, got this |
| 17:39.18 | danfromuk | channel.c: Unable to find a codec translation path from (alaw) to (g729) |
| 17:39.36 | danfromuk | but g729 show licenses shows 7, and there are no active calls. |
| 17:39.59 | Greenlight | Im not sure why we're seeing alaw mentioned at all if we're using g729 |
| 17:40.02 | danfromuk | The dialplan was working fine in 1.8. This started occuring since upgrading to 11 |
| 17:40.08 | Greenlight | Unless that Local channel is doing something odd |
| 17:40.26 | Greenlight | Does that Local answer the call before Dial ? |
| 17:41.05 | danfromuk | I think Client's Peer <-> Asterisk is g729. And Asterisk <-> Wholesale Provider is alaw |
| 17:41.19 | Greenlight | Right |
| 17:41.25 | Greenlight | So, you should be transcoding |
| 17:41.45 | Greenlight | What does "core show translation" show ? |
| 17:42.11 | danfromuk | Can't see g729 listed |
| 17:42.14 | przerull | @greenlight and igcewieling, yeah tcpdump looked pretty good. back to the drawing board. Thanks a lot |
| 17:42.25 | navaismo | module show like codec will show you the use count |
| 17:42.37 | igcewieling | danfromuk: what is the output of "g729 show version" |
| 17:42.37 | danfromuk | http://pastebin.com/AtL14hw9 |
| 17:42.50 | danfromuk | Digium G.729A Module Version 11.0_3.1.5 (optimized for barcelona_64) |
| 17:43.13 | Greenlight | I don't see g729 |
| 17:43.14 | Greenlight | ... |
| 17:43.41 | danfromuk | Me neither, but the modules definetly loaded because the cli has the g729 commands |
| 17:43.56 | danfromuk | codec_g729a.so Digium G.729 Annex A Codec (optimized fo 0 |
| 17:44.08 | danfromuk | 0 in the use count column |
| 17:44.17 | Greenlight | Looks like Asterisk is unaware of it being able to transcode |
| 17:44.38 | Greenlight | Call Digium support |
| 17:44.44 | mjordan | What does the Status column say when you doe a module show like codec_g729? |
| 17:45.38 | danfromuk | mjordan: status column? There is no status column. Here is the output. http://pastebin.com/8VcTLa08 |
| 17:46.01 | mjordan | blech. Forgot we added that in trunk :-P |
| 17:46.23 | Greenlight | Sounds like it may have been useful :) |
| 17:46.32 | mjordan | It is for situations like this |
| 17:46.52 | Qwell | if it shows 7 licenses, it's loaded |
| 17:47.01 | Qwell | started/whatever |
| 17:47.05 | danfromuk | Let me restart asterisk. I think i added the g729 module by doing module reload. Maybe a full restart will sort it. |
| 17:47.11 | Greenlight | WHy would be not see a translation path then ? |
| 17:47.24 | Greenlight | danfromuk: Good call.. |
| 17:47.33 | Qwell | If it had no licenses available when it tried to calculate the paths, it wouldn't have been able to. |
| 17:47.56 | Greenlight | Oh, we need to do a recalc if there's a new codec been added ? |
| 17:48.05 | danfromuk | Ok, after a restart, i can see g729 in core show translations. |
| 17:48.16 | Greenlight | SOunds promising |
| 17:48.31 | danfromuk | So module reload doesnt reload the paths |
| 17:48.48 | Greenlight | Yea, sounds that way |
| 17:49.40 | *** join/#asterisk tapout (~tapout@unaffiliated/tapout) |
| 17:49.55 | danfromuk | Ok, thanks. I've asked the client to retest and will see what the results are. |
| 17:49.58 | danfromuk | Thanks for the help. |
| 17:50.08 | Greenlight | Hope it works now! |
| 17:50.08 | orthoguy | I have a potential security issue that in all the internet searching, I haven't found an answer to. How can an international number get through my SIP trunk (Cox Cable's Edgemark 4550) and try to be an extension on my inbound context. It's almost like that number is registered as a DID or something. |
| 17:51.06 | Greenlight | orthoguy: I don't really understand the question... it can't unless you've something wrong in your dialplan I guess |
| 17:51.10 | igcewieling | orthoguy: unlikely as your carrier (Cox) is unlikely to allow you direct connection using your own equipment instead of theirs. |
| 17:51.28 | danfromuk | orthoguy: can you pastebin any cli output? |
| 17:51.57 | orthoguy | I'm getting interesting notices in /var/log/messages |
| 17:53.10 | orthoguy | chan_sip.c: Call from 'mynumber' (192.168.0.60:5050) to extension '01197599470065' rejected because extension not found in context 'cox_in' |
| 17:53.28 | orthoguy | then 15 seconds later it's the same thing |
| 17:53.38 | orthoguy | but with a + infront of the international number |
| 17:53.42 | danfromuk | It appears that 192.168.0.60 is dialling 01197599470065 |
| 17:53.47 | orthoguy | then a 9 |
| 17:53.53 | danfromuk | Is 192.168.0.60 within your network? |
| 17:53.53 | Greenlight | And.... |
| 17:53.56 | orthoguy | *.60 is the EdgeMark from cox |
| 17:54.08 | Greenlight | So, what's the issue here? |
| 17:54.30 | danfromuk | Have you spoken to cox to see if they can explain where the calls are coming from? |
| 17:54.42 | orthoguy | I guess I'm not sure why the edgemark is passing the internal number to my asterisk box |
| 17:54.44 | danfromuk | From the sounds of it I dont think its an asterisk issue. |
| 17:54.57 | orthoguy | I have called multiple times... they're new to SIP or something |
| 17:55.00 | Greenlight | 01197599470065 <-- that's not your number ? |
| 17:55.04 | orthoguy | It's been a frustrating experince |
| 17:55.06 | orthoguy | NOOOOO |
| 17:55.07 | orthoguy | it's not |
| 17:55.33 | Greenlight | ANd the call is from "mynumber"... hmm.. sounds like a misconfig |
| 17:55.40 | Greenlight | Grab a SIP trace and see what it's actually passing you |
| 17:55.47 | [TK]D-Fender | Someone is attempting to call out via your server |
| 17:55.52 | danfromuk | I'm confused. What exactly is an EdgeMark? |
| 17:55.57 | [TK]D-Fender | a SIP PROXY |
| 17:56.14 | orthoguy | It's a black box Cox put in my rack and said not to touch ;) |
| 17:56.16 | [TK]D-Fender | This isn't a call from "his provider" .... it's just passing his LOCAL PROXY\ |
| 17:56.41 | Greenlight | Oh, I thought this was a private network... if it's open to internet that would make sense |
| 17:56.43 | danfromuk | So its not necessarily coming from 192.168.0.60? |
| 17:56.46 | igcewieling | gawd, I'd rather have a limb amputated |
| 17:57.24 | igcewieling | A back box sip proxy. I think those come from the 4th circle of Dante's hell. |
| 17:57.30 | orthoguy | hahahahaha |
| 17:57.47 | danfromuk | orthoguy: turn off the EdgeMark and see if you get the same messages. |
| 17:58.17 | danfromuk | orthoguy: If they stop, then its appears that the EdgeMark is insecure and letting someone send calls to your asterisk box. |
| 17:58.25 | [TK]D-Fender | ... |
| 17:59.35 | orthoguy | danfromuk: that was my guess |
| 17:59.50 | orthoguy | they only come in every now and again |
| 18:00.21 | orthoguy | but they come in clusters |
| 18:00.33 | orthoguy | 01197599470065 |
| 18:00.35 | orthoguy | +01197599470065 |
| 18:00.40 | orthoguy | 901197599470065 |
| 18:00.44 | orthoguy | 701197599470065 |
| 18:00.49 | orthoguy | 801197599470065 |
| 18:00.55 | orthoguy | 001197599470065 |
| 18:01.05 | [TK]D-Fender | orthoguy: PASTEBIN <- Do not flood in here |
| 18:01.07 | [TK]D-Fender | ~pb |
| 18:01.08 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 18:01.09 | [TK]D-Fender | ^^^ |
| 18:01.12 | danfromuk | Is it always ending in 70065? |
| 18:01.14 | orthoguy | sorry |
| 18:01.19 | danfromuk | Or does the main part change? |
| 18:01.25 | WIMPy | Not that that list ist of particular interest. |
| 18:01.41 | *** join/#asterisk jetlag (~jetlag@pool-71-168-251-247.cmdnnj.east.verizon.net) |
| 18:01.46 | WIMPy | But it's interesting what kind of prefixes these guys try. |
| 18:01.47 | orthoguy | the 70065 stays the same |
| 18:01.52 | orthoguy | but... |
| 18:02.04 | Greenlight | Yea, they're poking around trying to find one that works |
| 18:02.04 | orthoguy | sometimes its a different number altogher |
| 18:02.08 | orthoguy | but it's the same pattern |
| 18:02.13 | orthoguy | with the end digits staying the same |
| 18:02.27 | WIMPy | Usually they try UK numbers. |
| 18:02.45 | danfromuk | orthoguy: If it stays the same, it could be a dialplan issue within the cox box. If it changes each time, then its more likely to be an abuse attempt. |
| 18:03.58 | igcewieling | orthoguy: those look like hack attempty |
| 18:04.21 | danfromuk | Either way, I would never trust a company to install a box in my rack if A) I didnt have total access to it, B) if I classed them as 'new to sip' |
| 18:05.00 | igcewieling | +975 – Bhutan |
| 18:05.26 | igcewieling | WIMPy: in my experience they try African numbers |
| 18:05.42 | Greenlight | We've seen a load in Gaza |
| 18:06.17 | danfromuk | On the Asterisk box, I would do a tcpdump on 192.168.0.60 to confirm the attempts are coming from the cox box. Then I'd send the cox box back and ask them for information on connecting asterisk directly to their sip trunks. |
| 18:06.36 | danfromuk | Greenlight: +1 for gaza. |
| 18:06.48 | igcewieling | Nah, tell them their box is trying to make international calls. |
| 18:06.56 | danfromuk | Its been a long time since i've seen any attempts though. |
| 18:07.02 | igcewieling | UNAUTHORIZED international calls. |
| 18:07.19 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 18:07.22 | igcewieling | At the company I work for (a CLEC and ITSP) that phrase will get our attention VERY quickly. |
| 18:07.40 | fullstop | I've been away from asterisk since 1.6.x, and I set up 11.5 over the weekend. I'm really impressed with how stable chan_motif is. Great work! |
| 18:10.54 | orthoguy | OK Sounds like its a cox issue and not here at this end. Thank you all very much for your help |
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