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| 01:18.46 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.18.5 (2017/12/22), Standard: 15.1.5 (2017/12/22); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 01:28.37 | avb | ups, my bad |
| 01:28.59 | avb | i meant master_channel but forgot to write it |
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| 04:33.31 | longus_catus | How do I get music on hold to work at a higher sampling rate than 8k, for use with codecs like OPUS, etc...? |
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| 06:44.55 | bllackjack | Can anyone help me out in changing the recording path directory? I'm able to change the directory path but I want to change the year/month/day convention and just want all the files saved in one folder! |
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| 06:51.31 | [TK]D-Fender | They go where you tell them to. |
| 06:59.53 | bllackjack | [TK]D-Fender: I'm able to change the directory but the files are still saved "custom-directory/year/month/day/file.wav" |
| 07:00.07 | bllackjack | Instead I want "custom-directory/file.wav" |
| 07:00.49 | [TK]D-Fender | Show a complete call |
| 07:02.32 | bllackjack | [TK]D-Fender: Couldn't get you, sorry! |
| 07:02.46 | [TK]D-Fender | pastebin a complete call where you're using this |
| 07:02.49 | [TK]D-Fender | ~pb |
| 07:02.49 | infobot | well, 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://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 07:04.15 | bllackjack | [TK]D-Fender: I just changed the default path from FreePBX portal. |
| 07:05.01 | [TK]D-Fender | You shouldn't be asking this in here then |
| 07:05.11 | [TK]D-Fender | It isn't just some singular line in an * config. |
| 07:05.19 | [TK]D-Fender | That entire system is built from the GUI |
| 07:05.44 | [TK]D-Fender | including the determination for their paths because their GUI depends on it to match up the recordings with CDR reporting,e tc |
| 07:06.08 | [TK]D-Fender | Remember that FreePBX owns your configs for the most-part, not you |
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| 07:40.58 | bllackjack | [TK]D-Fender: Can you move me in a direction to make that change? I plan on leaving FreePBX soon and want to handle the saving of wavs file customly. |
| 07:41.27 | [TK]D-Fender | Go name them yourself |
| 07:41.36 | [TK]D-Fender | There isn't a "tip" for this |
| 07:42.23 | [TK]D-Fender | read monitor()'s instructions |
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| 09:34.12 | babak | Hi, Is there a way we send real caller id to provider but ask provider to not send caller id to destination ? ( calller id restriction) |
| 09:47.52 | DanQuinney | babak: yes, either via a PAID and Privacy header, or RPID header |
| 09:48.26 | DanQuinney | As part of your INVITE |
| 09:50.29 | babak | DanQuinney: thx |
| 09:51.51 | DanQuinney | I send both to upstreams, but only one is needed |
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| 16:48.15 | lvlinux | for purposes of AMI, what is the mailbox number variable called? I'm wanting to use it in a "Getconfig" action in AMI and only bring me the line of a certain mailbox using the "Filter:" parameter. |
| 16:53.32 | [TK]D-Fender | go grab one without the filter and read it |
| 17:01.47 | lvlinux | [TK]D-Fender: yes I did that first of course, but it doesn't specify the names of the variables it is retrieving, just the lines with line/category numbers. |
| 17:02.58 | lvlinux | It brings me a bunch of lines like this: |
| 17:02.59 | lvlinux | Line-000000-000034: 1667=-1234,PurpleMartin,,,attach=yes|tz=eastern |
| 17:03.16 | lvlinux | Line-000000-000035: 1687=-1234,Oriole,,,attach=yes|tz=eastern |
| 17:03.41 | [TK]D-Fender | well you've got what you've got. |
| 17:03.46 | lvlinux | But I want to only get the line with mailbox 1687. |
| 17:04.38 | lvlinux | I tried specifying "Filter: mailbox=1687" and "Filter: mailbox=^1687$" and it says there aren't any results for those. |
| 17:06.13 | lvlinux | So I'm thinking that "mailbox" isn't the variable name or something. |
| 17:07.08 | [TK]D-Fender | If you dumped it without a filter you'd see what is actually included and not. |
| 17:07.41 | [TK]D-Fender | This isn't a "think" scenario. It's either there or it isn't and a single test should tell you concretely if it's a yes or a no |
| 17:08.19 | lvlinux | I dumped it without a filter and got lines like I said above. |
| 17:08.30 | [TK]D-Fender | then "think" isn't part of it. |
| 17:09.00 | [TK]D-Fender | that's what you get. Maybe you'll have to do a bit more parsing or filter on something else |
| 17:11.37 | lvlinux | The filter parameter says it is "A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered." |
| 17:12.29 | lvlinux | Oh, it says "categories" doesn't it, not "lines". Phooey. |
| 17:12.52 | lvlinux | i guess I'll have to parse them all and select the one i want in my code then. |
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| 20:43.28 | Brian1001 | hi everyone, does someone knows what this error means :ERROR[4605]: chan_pjsip.c:2227 in request: Failed to create outgoing session to endpoint 'phone6' |
| 20:43.46 | seanbright | are there additional messages? |
| 20:44.25 | seanbright | probably no AORs for that endpoint... but that is only a guess |
| 20:44.34 | Brian1001 | i cant see them sadly problem is that i can't call out when i connect my phone to an Asterisk box of my provider |
| 20:44.43 | Brian1001 | when i use my local asterisk machine, everything works great |
| 20:44.52 | Brian1001 | but they're sure that my phone is the problem :P |
| 20:45.52 | Brian1001 | ERROR[9978]: chan_pjsip.c:2227 in request: Failed to create outgoing session to endpoint 'Phone2' |
| 20:46.15 | seanbright | what version of Asterisk? |
| 20:47.30 | Brian1001 | 1.14 they said |
| 20:47.36 | Brian1001 | i can maybe check it in the web somewhere |
| 20:48.32 | seanbright | that cannot be correct |
| 20:48.55 | Brian1001 | checking if i can find the exact version |
| 20:49.01 | seanbright | you don't have access to the box? |
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| 20:49.29 | Brian1001 | i do , but they have a webinterface which sucks |
| 20:49.35 | Brian1001 | they try to hide a lot of info |
| 20:49.40 | seanbright | gotcha |
| 20:51.47 | Brian1001 | 14.7.2 |
| 20:52.19 | Brian1001 | that seems to be the asterisk version |
| 20:52.38 | seanbright | yes. that is almost the latest version of Asterisk 14.x |
| 20:53.12 | seanbright | so if i understand correctly, your provider has given you an Asterisk box with a web-based GUI |
| 20:53.22 | seanbright | you have a phone you are trying to register with that box |
| 20:53.37 | seanbright | and you are not able to call that phone via the Asterisk server once it is registered? |
| 20:53.38 | Brian1001 | yes exactly |
| 20:54.03 | Brian1001 | and i use their trunk on the asterisk box |
| 20:54.06 | seanbright | who is your provider? PM if you need to keep that a secret |
| 20:54.28 | Brian1001 | may i indeed PM you ? |
| 20:54.42 | seanbright | i just said so, so yes |
| 20:55.57 | seanbright | ok, do you have the ability from their interface to increase the logging verbosity? |
| 20:56.09 | Brian1001 | yes |
| 20:56.23 | seanbright | you should do that |
| 20:56.30 | seanbright | debug if possible |
| 20:56.40 | seanbright | and then pastebin the call attempt |
| 20:56.40 | seanbright | ~pb |
| 20:56.41 | infobot | from memory, 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://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 20:56.45 | Brian1001 | i will try to do that |
| 21:00.06 | Brian1001 | Thank You for helping seanbright |
| 21:00.16 | seanbright | sure thing |
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| 21:28.36 | jfindley | is there any conflict between fastagi and asyncagi? |
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| 21:42.36 | seanbright | jfindley: conflict in what way? |
| 21:43.52 | jfindley | if I'm inside a fastagi script and AMI attempts to execute an AGI action (e.g., playback), will that break everything or do they play well together? |
| 21:44.41 | seanbright | i don't see why that wouldn't work. are you having problems or just trying to determine if you will? |
| 21:45.06 | jfindley | having problems and wondering if that's where the issue is coming from. |
| 21:49.07 | seanbright | i think i am confused. how are you executing both FastAGI and AsyncAGI on the same channel at the same time? |
| 21:51.45 | jfindley | fastagi happens through the dialplan, ami happens externally. I'm trying to achieve the ability to play arbitrary sounds on arbitrary channels at arbitrary times |
| 21:52.04 | Kobaz | i have an odd thing going on here |
| 21:52.10 | Kobaz | [2017-12-27 14:34:09.656] VERBOSE[8735] channel.c: [2017-12-27 14:34:09.656] -- <SIP/isoft-sr-in-1-0000102f> Doing SoftHangup (0x7f554c4437d8) (cause: 2) |
| 21:52.23 | Kobaz | but the channel continues on, and later hangs up at 14:38 after having a conversation |
| 21:52.49 | seanbright | jfindley: so you aren't using AsyncAGI. you're using AGI(agi://...) and then using ControlPlayback via AMI? something along those lines |
| 21:53.00 | jfindley | yep |
| 21:53.02 | seanbright | Kobaz: a soft hangup is not a hangup |
| 21:53.08 | Kobaz | cause 2 is no route found? |
| 21:54.00 | seanbright | jfindley: while AGI() is running it "owns" the channel, so you may not be able to do what you want |
| 21:54.16 | seanbright | jfindley: if ARI is an option for you, that might be a better approach |
| 21:54.22 | Kobaz | seanbright: what's involved in cause 2 and why would the channel not go away? |
| 21:54.43 | jfindley | seanbright: Thanks, I'll see if that'll work for this case. |
| 21:55.52 | seanbright | Kobaz: i may be talking out of my ass |
| 21:55.59 | seanbright | and i have to go away for a few |
| 21:57.37 | file | a soft hangup may or may not be a true hangup, it is also used as an interrupt mechanism, 2 is probably 1 << 1 which is used to do an "async goto" |
| 21:57.56 | Kobaz | checking the sip dump, i have some reinvites going on |
| 21:58.18 | Kobaz | i do have a Redirect() in progress as well |
| 21:58.28 | Kobaz | okay so the async goto sounds reasonable |
| 22:00.15 | Kobaz | yeah okay |
| 22:00.35 | Kobaz | 2017-12-27 14:34:09.646, [AgiDaemon (Core) Manager] (26937) <SIP/isoft-sr-in-1-0000102f> (HoldCallerAfterIntercomXfer) We [CallLogID: 14067] are now holding until <SIP/100-00001030> picks us up |
| 22:00.38 | Kobaz | from my own logging |
| 22:00.48 | Kobaz | that sends the Redirect |
| 22:01.17 | Kobaz | okay whew, the softhangup was really confusing |
| 22:02.33 | Kobaz | now i gotta see what happened to SIP/100-00001030 |
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