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| 01:52.20 | nafg | Hi, I posted on the mailing list last week and still haven't gotten a reply: http://lists.digium.com/pipermail/asterisk-app-dev/2017-October/000893.html |
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| 02:21.24 | [TK]D-Fender | that is an ancient release and not supported. |
| 02:21.26 | [TK]D-Fender | upgrade |
| 02:21.38 | [TK]D-Fender | Asterisk certified/13.1-cert4, Copyright (C) 1999 - 2014, Digium, Inc. and |
| 02:21.38 | [TK]D-Fender | others. |
| 02:21.44 | [TK]D-Fender | Please look at the date |
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| 02:54.01 | Spajjie | Hi |
| 02:54.42 | Spajjie | Does anybody have knowledge of Custom Modules? I'm trying to upload a edited HelloWorld module back onto my test system.. it's a tar.gz file and when I upload my file I get a "Error from tar" but nothing more specific. |
| 02:56.59 | [TK]D-Fender | A tar fail isn't a "module" |
| 02:57.05 | [TK]D-Fender | that's the same as saying a "zip file" |
| 02:57.12 | [TK]D-Fender | it's an archive format which means nothing |
| 02:57.19 | [TK]D-Fender | * modules are compiled .so 's |
| 02:58.57 | Spajjie | Not 100% sure what you're trying to tell me.. but by rights if I download the HelloWorld example and tar.gz the directory structure and upload it. it should see it as a module? |
| 02:59.07 | Samot | What "HelloWorld" example? |
| 02:59.15 | Spajjie | https://github.com/FreePBX/helloworld |
| 02:59.20 | Samot | OK |
| 02:59.21 | Samot | Wrong room |
| 02:59.23 | Samot | #freepbx |
| 02:59.30 | Spajjie | Sorry chaps.. you're right |
| 02:59.34 | Samot | Even more specific #freepbx-dev |
| 02:59.43 | Spajjie | Thank you - my bad. |
| 03:03.08 | Samot | Uhm. |
| 03:03.13 | Samot | That module is over 4 years old. |
| 03:03.25 | Samot | It no longer applies to the way modules are done. |
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| 05:20.01 | nafg | [TK]D-Fender: I just tried asterisk 14, same thing |
| 05:46.42 | [TK]D-Fender | post an issue for it on the tracker |
| 05:46.57 | [TK]D-Fender | The mailing list is not the way to go when dealing with an issue like this |
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| 08:35.52 | wyoung | .o/ |
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| 12:22.47 | gedia | hello guys. I need to have asterisk send an in-dialog re-invite with a=sendonly in the SDP to test functionality of an RTP proxy, but putting a call on hold on a phone results in the B-leg transmitting MOH inband without a reinvite. Is there any way to enforce a re-invite to be sent on B-leg? |
| 12:24.04 | file | that is a feature available in chan_pjsip as of Asterisk 15 |
| 12:24.13 | file | it's configured using the "moh_passthrough" option on the endpoint |
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| 12:32.12 | gedia | file: oh ok... thanks! |
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| 17:25.24 | nafg | In a FastAGI app connected via SIP, how can I get the phone number of "the other party" |
| 17:25.34 | nafg | If it's an incoming call I can use caller id |
| 17:26.32 | nafg | If it's an outgoing call (set up via an Originate), how can I access the destination number? |
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| 17:27.12 | nafg | [TK]D-Fender: were you saying to post an issue on the issue tracker to me? |
| 17:27.22 | nafg | (last night) |
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| 17:39.43 | [TK]D-Fender | yes |
| 17:40.05 | [TK]D-Fender | AGI doesn't have "another party |
| 17:40.12 | [TK]D-Fender | It is not a background thing to begin with |
| 17:40.22 | [TK]D-Fender | it is a blocking alternative to standard dialplan |
| 17:40.44 | sawgood | Hi: Using Asterisk 13.17.2: in order to use HEP (do) we need to "add" a module 1st with makemenuselect? |
| 17:41.34 | [TK]D-Fender | <nafg> If it's an outgoing call (set up via an Originate), how can I access the destination number? <- depends exactly when and how you are trying to geet to it. You need to be clearer about how "in the mix" this has to be determined |
| 17:42.51 | nafg | [TK]D-Fender: idk what "in the mix" means. What do I need to be clearer about? |
| 17:43.21 | [TK]D-Fender | Exactly how you are processing this call and at what stage you are trying to get this information |
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| 17:45.06 | nafg | [TK]D-Fender: I send an OriginateAction via AMI, with Application set to Agi and Data set to the agi:// URL of the handler |
| 17:45.53 | nafg | Then, at various points during the call, for analytics reasons, I want to "log" the number that was originated to |
| 17:46.34 | nafg | I get that this is not part of the AGI information per se, but surely it's in asterisk somewhere? Maybe something I could GET FULL VARIABLE on? |
| 17:47.05 | nafg | I just checked and it's not anywhere in the initial AGI environment |
| 17:47.08 | [TK]D-Fender | lIt may be in the CDR. |
| 17:47.08 | nafg | AFAICT |
| 17:47.19 | nafg | How do I access the CDR? |
| 17:47.26 | [TK]D-Fender | But rather than thinking that way... set a variable in your Originate with that value so you have it from the start |
| 17:49.16 | nafg | [TK]D-Fender: hah, I just realized I already do. How do I access it? It's called DEST_NUM |
| 17:49.29 | nafg | I need to do a GET VARIABLE? |
| 17:49.52 | nafg | Also is there any simple way to tell if I'm in an inbound or outbound call? |
| 17:49.53 | [TK]D-Fender | just like any other... |
| 17:51.24 | nafg | Any other what? I'm not exactly an asterisk expert. So far I see there is the request "environment", and then there are things you have to GET VARIABLE on, and I don't know if there's anything else |
| 17:51.46 | [TK]D-Fender | that is exactly it |
| 17:52.03 | [TK]D-Fender | get variable = Gets the value of an * channel variable |
| 17:52.25 | nafg | Ok great thanks |
| 17:52.51 | nafg | [TK]D-Fender: any way to tell if it's inbound or outbound? Or should I just check if DEST_NUM exists? |
| 17:54.01 | [TK]D-Fender | If what's Inbound? |
| 17:54.15 | [TK]D-Fender | you are calling OUT with an originate. |
| 17:54.30 | [TK]D-Fender | and then dumping them into an AGI (instead of dialplan) |
| 17:54.39 | [TK]D-Fender | How is there any room for confusion? |
| 17:56.10 | nafg | [TK]D-Fender: WDYM, an AGI script can run for an incoming call and for an outgoing call |
| 17:56.20 | nafg | So in the AGI script I don't automatically know |
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| 17:59.41 | [TK]D-Fender | .. |
| 17:59.50 | [TK]D-Fender | YOu aare starting a call OUT using Originate |
| 17:59.58 | [TK]D-Fender | What part of that says "inbound" at all? |
| 18:00.22 | [TK]D-Fender | Where does the idea of "inbound" even exist? |
| 18:00.40 | [TK]D-Fender | What other kind of call are you talking about? |
| 18:01.19 | [TK]D-Fender | If you are processing an incoming call and calling an AGI that is just another channel with it's own CID & properties. |
| 18:01.19 | Penguin | [TK]D-Fender: its |
| 18:02.20 | [TK]D-Fender | So when using Originate you are setting a variable for the number to have it conveniently accessible... that same way you can know that you in there because of starting from that Originate |
| 18:02.35 | [TK]D-Fender | Or you could have 2 different scripts |
| 18:03.12 | [TK]D-Fender | You could set a var in just the "inbount call dialplan before hitting the AGI. Or pass the AGI a parameter |
| 18:03.22 | [TK]D-Fender | Or something else... |
| 18:05.31 | nafg | [TK]D-Fender: the AGI script is not the code that called originate. The AGI script can be reached from the dialplan from incoming AND from originating |
| 18:05.44 | [TK]D-Fender | I've answered this above |
| 18:06.15 | nafg | I think the answer was there's no built in way |
| 18:06.49 | nafg | (yes there are workarounds, like checking if DEST_NUM is set) |
| 18:07.06 | [TK]D-Fender | nothing about either says "outgoing" |
| 18:07.13 | [TK]D-Fender | a channel is created and lands in diallpan. |
| 18:07.43 | [TK]D-Fender | The dialplan knows nothing more than the callerid, the targeted exten (if that was even a thing). |
| 18:07.48 | [TK]D-Fender | and that's about it |
| 18:07.55 | [TK]D-Fender | it's just "a channel" |
| 18:08.03 | nafg | with a bunch of variables in it |
| 18:08.08 | [TK]D-Fender | ... or not\ |
| 18:08.27 | [TK]D-Fender | So call your script differently. Or set some variables. Or split your scripting up. |
| 18:08.42 | nafg | Yeah, that stuff I can work out |
| 18:09.06 | [TK]D-Fender | agi(Myscript,INCOMING) |
| 18:09.18 | [TK]D-Fender | pass as a parm. |
| 18:09.20 | [TK]D-Fender | or whatever |
| 18:09.26 | nafg | [TK]D-Fender: about the popping in my audio files, were you saying to file a bug report? I think I misunderstood something |
| 18:09.33 | [TK]D-Fender | Or look for that dest you explicitly have to set |
| 18:09.39 | nafg | Yeah thanks |
| 18:09.45 | [TK]D-Fender | Yes, I'm saying to file a bug report |
| 18:09.56 | [TK]D-Fender | if you want stuff actually fixed, the ML isn't the place to go. |
| 18:10.07 | nafg | I don't think it's a bug in asterisk |
| 18:10.29 | nafg | I mean if it is a bug in asterisk what would the bug be? |
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| 18:16.46 | [TK]D-Fender | Is there popping on a local playback? |
| 18:16.54 | [TK]D-Fender | on a PC directly? |
| 18:17.03 | [TK]D-Fender | On local LAN device? |
| 18:19.22 | fauxalliance | the docker image? |
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| 18:19.27 | fauxalliance | that _bug_? |
| 18:21.41 | nafg | [TK]D-Fender: there is popping even from a local asterisk |
| 18:21.58 | nafg | fauxalliance: what? |
| 18:22.38 | nafg | [TK]D-Fender: i.e. a locally running SIP phone connecting to a local asterisk (in docker) via localhost |
| 18:22.54 | fauxalliance | sorry, it' was a bug report earlier in scrollback, about a docker image audio popping and spritzing |
| 18:23.03 | fauxalliance | that clearly wasn't a bug |
| 18:24.02 | nafg | fauxalliance: not sure if you're talking about my earlier posting. I'm having audio popping, docker has nothing to do with it |
| 18:24.22 | nafg | it just happens to be running in docker |
| 18:24.30 | nafg | https://chavrusa.tk/admin/speech#Prompts |
| 18:24.44 | nafg | oops that was a copy-paste error |
| 18:25.07 | nafg | I mean was an error that my clipboard got pasted into the chat |
| 18:25.31 | nafg | or something like that |
| 18:30.36 | [TK]D-Fender | is it in the audio files? Does it occur on a non-Docker install to a non-Docker client? |
| 18:31.22 | fauxalliance | yes, iirc, on both WAV and sln |
| 18:32.43 | fauxalliance | but I ain't seen nothing |
| 18:32.49 | fauxalliance | grammaire anglaise, meh. |
| 18:52.59 | tafa2 | does anyone use anveo? |
| 18:53.23 | [TK]D-Fender | ? polls |
| 18:53.29 | [TK]D-Fender | ~polls |
| 18:53.29 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
| 18:54.50 | tafa2 | does anyone use anveo? and have a personal opinion on how to improve the call quality of prime routes? Do you find that service tier ordering works well long term? |
| 18:57.14 | tafa2 | I'm also looking for a better outbound provider for EMEA destinations - would anyone have any recommendations? |
| 18:57.21 | tafa2 | Prime routes |
| 18:57.34 | nafg | [TK]D-Fender: I found one audio file that I get a slightly similar effect in Audacity, but the same file when played via play (from sox package) or aplay (from alsa-utils package) is fine |
| 18:58.11 | [TK]D-Fender | Sounds like you should do another encoding job on something cleaner |
| 19:01.26 | nafg | [TK]D-Fender: there are definitely a plenty of quality issues. But I don't know how to avoid it if I don't know what's causing it |
| 19:01.59 | nafg | I record a lot of short phrases and stitch them together (play them in order) in code |
| 19:02.08 | nafg | so I have a lot of prompts |
| 19:05.43 | fauxalliance | I have a lot of rabid weasels. |
| 19:16.51 | *** join/#asterisk tzafrir (~tzafrir@212.29.194.37) |
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| 19:58.46 | chris349 | Is there any way to undo issuing the command core restart gracefully? |
| 20:03.08 | [TK]D-Fender | try hitting tab and look at the options |
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| 21:49.39 | *** join/#asterisk KNERD (~KNERD@2604:a880:1:20::9b:4001) |
| 21:52.06 | KNERD | WHy is Digium depending on Third Party websites to download content it may need? |
| 21:52.35 | KNERD | checking for embedded pjproject (may have to download)... configuring [pjproject] Downloading https://raw.githubusercontent.com/asterisk/third-party/master/pjproject/2.6/pjproject-2.6.tar.bz2 t |
| 21:52.41 | Samot | Because those third parties are the ones contributing. |
| 21:52.47 | KNERD | Annnd...it's down |
| 21:52.48 | Samot | PJProject is not Asterisk's. |
| 21:53.32 | file | we mirrored PJSIP on Github, as the PJSIP downloads site was slow/unreliable |
| 21:53.55 | KNERD | Well, if they are going to include a project, maybe have an aletrnative mirror |
| 21:54.30 | KNERD | okay..I will manually install it then I guess |
| 21:54.50 | file | it's working here, but Github uses a CDN so perhaps the specific edge node you hit had a problem |
| 21:55.11 | *** join/#asterisk _r0ck_p3arl_ (~admin@203.149.66.146) |
| 21:55.15 | Samot | I just clicked on that link and it downloaded. |
| 21:55.18 | Samot | 4.7MBs |
| 21:55.25 | KNERD | I tried it from 2 different IP addresses from 2 different ISP in 2 different countries |
| 21:56.05 | tafa2 | In London - it works |
| 21:56.18 | KNERD | Hm...however. they are using same DSN service to resolve...Googl..thanks |
| 21:56.32 | tafa2 | 4.7MBs |
| 21:58.10 | tafa2 | https://we.tl/zVl259ejv8 there ya go |
| 22:01.41 | KNERD | Well it's not a Google DNS issue as I thought.only one is using Google DNS |
| 22:02.09 | KNERD | thanks |
| 22:02.46 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
| 22:04.46 | KNERD | Yeah when the version states stated PJSIP was now included with Asterisk, I took that literraly, not a thrid party download |
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| 22:16.23 | KNERD | it's just one thing after another today. Now PJSIP won;t compile |
| 22:17.00 | KNERD | pjproject-2.6/third_party/lib/libwebrtc.so: undefined reference to `__gxx_personality_v0' |
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| 22:18.22 | Samot | So. Many. Zingers. Must. Resist. |
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| 22:23.16 | KNERD | Yes..Zapped your Zinger |
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| 22:55.37 | *** join/#asterisk clarjon1 (~clarjon1@unaffiliated/clarjon1) |
| 22:56.09 | clarjon1 | Ok, seems i've buggered my personal pbx install. not sure how |
| 22:56.34 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:56.59 | clarjon1 | From what i can see... it's not connecting to the trunk properly, i think. |
| 22:57.23 | clarjon1 | The stats say that the trunk is online in freepbx, but in flowroute, it says nothing's connected, and i get a "not in service" message when i try calling my DID |
| 22:57.31 | [TK]D-Fender | "maybe have a misconfigured trunk" != "buggered PBX Install" |
| 22:57.50 | clarjon1 | Well, I've not made changes to the trunk config. |
| 22:58.11 | clarjon1 | I have applied updates to FreePBX |
| 22:58.14 | [TK]D-Fender | that "in service" is nothing more than QUALIFY simply getting answered. |
| 22:58.22 | [TK]D-Fender | it does not mean you've REGISTERED to them at all |
| 22:58.47 | [TK]D-Fender | Go look at the registry status and the SIP traffic for those attempts, etc |
| 22:59.09 | clarjon1 | Where would I find that? |
| 22:59.26 | clarjon1 | i'm not seeing anything popping out as "registry status" in the dropdown menus in freepbx |
| 22:59.51 | [TK]D-Fender | * CLI |
| 22:59.57 | [TK]D-Fender | GUI is meaningless at this point |
| 23:00.04 | clarjon1 | ok |
| 23:00.44 | [TK]D-Fender | "not in service doesn't mean much yet either |
| 23:00.47 | [TK]D-Fender | place a call and show us |
| 23:01.13 | [TK]D-Fender | That could easily mean the trunk is fine and you've screwed up an inbound route |
| 23:01.21 | clarjon1 | [2017-10-17 23:00:32] NOTICE[2377] chan_sip.c: -- Registration for '[ID_REMOVED]@sip.flowroute.com' timed out, trying again (Attempt #12) |
| 23:01.28 | [TK]D-Fender | that's a bad sign |
| 23:01.32 | [TK]D-Fender | PASTEBIN <---- |
| 23:01.49 | clarjon1 | pastebin what exactly? |
| 23:02.13 | clarjon1 | huh interesting |
| 23:02.23 | clarjon1 | i'm also getting attempted calls coming in from other sip servers |
| 23:02.50 | clarjon1 | i AM on a dynamic IP. i wonder if a previous user on my ISP had a sip setup that was vulnerable, and got used for spamming calls |
| 23:03.08 | [TK]D-Fender | Show the failed call attempt |
| 23:03.13 | [TK]D-Fender | <[TK]D-Fender> place a call and show us |
| 23:03.34 | clarjon1 | oh, nothing shows up |
| 23:03.41 | clarjon1 | when calling in |
| 23:03.44 | clarjon1 | i'll try calling out, one sec |
| 23:03.48 | [TK]D-Fender | enable SIP debug and show the failed registration attempt |
| 23:03.58 | [TK]D-Fender | "sip set debug on" <- |
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| 23:04.02 | clarjon1 | can i get the link to those instrucitons again? |
| 23:05.16 | clarjon1 | brb |
| 23:09.24 | [TK]D-Fender | heads out shopping |
| 23:10.50 | fauxalliance | lola, gehst du einkaufen, braucht ich shampoo. |
| 23:12.39 | fauxalliance | my cousin said " VA te faire foute" |
| 23:18.46 | *** join/#asterisk clarjon1 (~clarjon1@unaffiliated/clarjon1) |
| 23:19.02 | clarjon1 | Issue resolved. |
| 23:19.42 | clarjon1 | I had a bunch of incoming sip connections from random IPs, so i figured that since i recently got the IP due to a power outage ( isp has dhcp for me ) which happened around the same time that the calls stopped working |
| 23:19.49 | clarjon1 | so, powercycle, fresh IP, connections work |
| 23:20.21 | clarjon1 | and i've let flowroute know that ip is blacklisted on their end, and that it's a DHCP assigned ip. |
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| 23:54.46 | Korolev | Hi, is anybody getting jittery audio when using pjsip? |
| 23:55.17 | Korolev | if I switch back to chan_sip, I get clear audio under the exact same conditions |
| 23:59.38 | file | there was a bug where chan_pjsip wouldn't switch to match the received codec, it could result in behavior like that, it's fixed in the 13.18 rc |