| 06:41.34 | *** join/#asterisk-dev infobot (ibot@c-174-52-60-165.hsd1.ut.comcast.net) |
| 06:41.34 | *** topic/#asterisk-dev is Take the March 2019 Asterisk User Survey! https://goo.gl/forms/xL1VUHRsf95saly13 -- Asterisk Development Discussion -=- http://www.asterisk.org/developers -=- Tier 2 and 3.14159265 support is in #asterisk -=- Check out our blog! blogs.asterisk.org -=- Follow on Twitter at @AsteriskDev |
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| 09:47.34 | jkroon | can someone please point me at patch submission process for dahdi-tools? doesn't seem to be on gerrit ... |
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| 11:32.47 | jkroon | https://gerrit.asterisk.org/c/asterisk/+/13309/4/res/res_rtp_asterisk.c#2987 so joshua (predictably so) was correct here. is there a way (without iterating the entire config file) to get "all values in a specific section for a given key"? |
| 11:34.12 | file | don't think so, no |
| 11:37.59 | jkroon | would it make sense to add a function to do that? |
| 11:41.12 | file | it's not really... designed... to be iterated in such a way |
| 11:42.02 | file | it's probably more efficient to just iterate the entire thing and skip any non-matching |
| 11:43.32 | jkroon | yea ... main/config.c looks like a nightmare ... |
| 11:45.47 | jkroon | right, so rather than iterating 8 times it's going to be much more efficient to restructure completely, iterate once, looking for matches, and dispatching accordingly. |
| 11:46.08 | file | yes. |
| 11:48.43 | jkroon | ok well ... so be it ... thanks. |
| 12:15.34 | gtjoseph | pchero_work: did you see the testsuite issue with your session termination patches? |
| 12:24.45 | pchero_work | gtjoseph: Hi, yes, I've just checked that. I'm digging it now. |
| 12:25.07 | gtjoseph | ok cool. just wanted to may sure you saw it :) |
| 12:25.38 | pchero_work | Thanks. :) |
| 12:35.22 | jkroon | https://issues.asterisk.org/jira/browse/DAHTOOL-85 - would anyone be able to comment if that patch makes sense? Also, regarding the correct submission procedure. |
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| 13:39.09 | jkroon | https://gerrit.asterisk.org/c/asterisk/+/13309 - would really appreciate a once-over and an opinion on this if someone has a few cycles. |
| 13:41.39 | file | everything that goes up on Gerrit gets reviewed |
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| 19:39.15 | *** join/#asterisk-dev Nir_S (~nirs@40.67.196.251) |
| 19:39.25 | Nir_S | good morning/afternoon all |
| 19:41.07 | DanJenkins[m] | hey |
| 19:42.53 | Nir_S | hey hey |
| 19:45.05 | Nir_S | Can anyone share some information about External Media in ARI ? |
| 19:45.16 | Nir_S | I seem to have a minor confusion about it |
| 19:45.56 | gtjoseph | Nir_S: What's up? |
| 19:46.08 | DanJenkins[m] | yes |
| 19:46.24 | DanJenkins[m] | what would you like to know |
| 19:46.28 | DanJenkins[m] | i can point you at an excellent reference application |
| 19:46.36 | DanJenkins[m] | (if i do say myself) |
| 19:46.51 | gtjoseph | :) |
| 19:49.38 | Nir_S | Hey Georg |
| 19:49.40 | Nir_S | George |
| 19:50.38 | Nir_S | I'm trying to understand if it is a bi-directional stream - namely, can I both read the RTP stream and play audio into it. In addition, how does NAT affect it - if at all ? |
| 19:50.52 | Nir_S | DJ, are you referring to the node application in the repo ? |
| 19:51.09 | DanJenkins[m] | yes its bidirectional |
| 19:51.12 | DanJenkins[m] | node application in my repo |
| 19:51.23 | DanJenkins[m] | https://github.com/nimbleape/asterisk-dialogflow-rtp-audioserver |
| 19:51.35 | DanJenkins[m] | and NAT ruins it completely |
| 19:53.22 | DanJenkins[m] | or at least it does for me.... I use the local port variable generated by externalMedia to be able to associate a media session in the other app (the one doing audio generation etc) to the media session itself |
| 19:53.38 | Nir_S | thanks DJ, thet's exactly what I needed |
| 19:53.41 | DanJenkins[m] | anything involving a nat changes what the app handling the media sees as the source port |
| 19:54.46 | DanJenkins[m] | it goes hand in hand with https://github.com/nimbleape/asterisk-dialogflow-ari-bridge |
| 19:54.55 | DanJenkins[m] | shout if you need anthing else :) |
| 19:55.16 | Nir_S | other than that, how is Covid19 treating everybody ? |
| 19:57.38 | gtjoseph | we're supposed to do "social distancing" now but I've always done that. :) |
| 19:57.58 | DanJenkins[m] | exactly! :D |
| 19:59.10 | Nir_S | ahh... I've already gathered many years ago that we'll all "selectively autistic" - so that's fine :-D |
| 19:59.17 | Nir_S | we're all |
| 20:00.28 | gtjoseph | yep :) |
| 20:01.37 | gtjoseph | In times like these it's fun to watch old "Yes Prime Minister" shows. |
| 20:02.18 | Nir_S | Dear lord |
| 20:02.30 | Nir_S | I think I hadn't heard that show name in 30 years |
| 20:02.31 | rmudgett | Is that something like "Are you being served?" |
| 20:02.45 | Nir_S | something like it, just a bit more brown nosed |
| 20:03.02 | gtjoseph | Actually it's more like "Are You NOT Being Served" |
| 20:03.24 | gtjoseph | great commentary on the political system |
| 20:03.58 | gtjoseph | and it's response to crisis |
| 20:04.36 | Nir_S | gt, please, do not mention the word "crisis" at this point - I think the world is messed up as is |
| 20:04.52 | gtjoseph | quite true |
| 20:04.55 | Nir_S | There are words about doing an 8 week shut-down to the country over here |
| 20:05.12 | gtjoseph | wow |
| 20:06.32 | gtjoseph | who knows, the world may be a better place if every country bans together and shares its toilet paper supplies. |
| 20:07.25 | gtjoseph | s/bans/bands/ |
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| 20:37.46 | Nir_S | DJ, GT, here is a funky question |
| 20:38.15 | Nir_S | Can External Media be connected to rtpengine ? |
| 20:39.36 | Nir_S | for example, issue a remote command to RTP engine that we're starting a new rtp session, then point the RTP stream over to rtpengine, so it can handle the various "codec transcoding if required" and "codec sanity and security" prior to letting someone push audio into Asterisk's RTP media buffers (which I believe is somewhat of a security risk) |
| 20:40.45 | gtjoseph | I can't say I've tried it but one issue that might crop up is the handling of dynamically assigned payloads. |
| 20:41.15 | DanJenkins[m] | Oh yeah that's a big gotcha ;) |
| 20:41.39 | DanJenkins[m] | RTP headers must include static RTP type |
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| 20:44.25 | sushiman | Hello, Is anybody able to compile asterisk with pjsip in a container running alpine? Thanks |
| 20:44.27 | DanJenkins[m] | It's why my project sends back slin16 8k and not slin16 16k |
| 20:44.44 | gtjoseph | sushiman: not that I know of. |
| 20:44.53 | gtjoseph | are you running into issues? |
| 20:46.09 | sushiman | yea. I am able to compile pjsip separately, but I don't know how to tell asterisk to use that. |
| 20:46.39 | gtjoseph | do you _need_ to compile pjsip separately? |
| 20:46.51 | DanJenkins[m] | Bundled pjsip not work? |
| 20:46.58 | gtjoseph | and which version of Asterisk? |
| 20:47.31 | sushiman | If I try to compile asterisk with the default pjsip it gives me an error like this https://pastebin.com/2g1Uh6Rn |
| 20:47.49 | sushiman | asterisk 17.3 |
| 20:48.00 | sushiman | bundle doesn't work |
| 20:48.42 | sushiman | I would prefer not to compile pjsip separately, but It doesn't seem to compile with asterisk |
| 20:48.48 | gtjoseph | well if you want to use non-bundled, you have to pass --without-pjproject-bundled to ./configure and make sure that your copy is installed in the correct system locations |
| 20:49.25 | sushiman | any way to test that asterisk can see the pjsip? |
| 20:49.58 | gtjoseph | if it doesn't allow you to set the res_pjsip stuff in menuselect, then it didn't. |
| 20:51.30 | gtjoseph | Also, if you're building with bundled, you can pass NOISY_BUILD=yes to 'make' to get more info on why it failed. |
| 20:52.58 | DanJenkins[m] | Looks like it isn't able to work out what kind of Linux it is to be able to go get the right stuff |
| 20:53.15 | DanJenkins[m] | Because you know.... Docker |
| 20:55.58 | sushiman | yea that seem to be the problem |
| 20:56.12 | sushiman | I will try to add the pjsip to the right places and test |
| 20:58.10 | sushiman | now I have a chan_sip error |
| 20:58.53 | sushiman | but I will try to debug this problem before I ask for help |
| 21:06.32 | DanJenkins[m] | If you're trying to get pjsip working I'd disable building chan_sip anyway |
| 21:08.25 | sushiman | how do I do that? --without-chan-sip? |
| 21:18.38 | snuff-work | sushiman: just disable via menuselect |
| 21:19.10 | snuff-work | i can't remember.. is app_voicemail still dependant on chan_sip in any way? |
| 21:30.44 | rmudgett | It never was dependent on chan_sip |
| 21:42.19 | sushiman | any help with this. I disabled chan_sip by commented the member tag with chan_sip in the menuselect-tree and I still having problems related to chan_sip https://pastebin.com/VHR6Q7Ky |
| 21:42.56 | sushiman | Although, I'm thinking it is maybe the compiler. |