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| 11:15.00 | skrusty | morning |
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| 15:10.31 | pabelanger | Well that stinks |
| 15:10.46 | pabelanger | asterisk does not display the actually 302 redirect SIP message on sip set debug on |
| 15:10.48 | pabelanger | I just get |
| 15:10.52 | pabelanger | Got SIP response 302 "Moved Temporarily" back from 198.41.29.45:19207 in log |
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| 15:32.17 | file | it doesn't differentiate unless you limit it |
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| 15:33.17 | pabelanger | I did have sip set debug peer x.x.x.x |
| 15:33.28 | pabelanger | will need to check code and see what is going on |
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| 15:53.58 | Stefan26 | hi americans |
| 15:54.10 | blitzrage | I'm only North American |
| 15:55.58 | Stefan26 | im not american at all I just assumed most of this channel was |
| 15:56.01 | Marquis42 | Yeah blitzrage is from America Junior. :P |
| 15:56.23 | opticron | * america's hat |
| 15:56.29 | Marquis42 | That too |
| 15:56.39 | blitzrage | lol, I was going to say that (America's hat) |
| 15:56.59 | blitzrage | we're like that awesome top hat that you wish you could wear in public |
| 15:57.04 | Marquis42 | lol |
| 15:57.16 | Marquis42 | You do know my statement was a Simpsons reference, right? |
| 15:58.14 | blitzrage | I will cut you |
| 15:58.25 | Marquis42 | :D |
| 15:58.27 | Stefan26 | Canada and US have never been at war? |
| 15:58.33 | Stefan26 | against each other |
| 15:58.55 | Marquis42 | I think at some point in the 19th century we were, but not in any major way IIRC. |
| 15:59.22 | Stefan26 | We (Sweden) have been at war with our neighbour (Denmark) 10 times throughout history |
| 15:59.41 | blitzrage | 1812 we were |
| 15:59.59 | blitzrage | Canada burned the whitehouse down |
| 16:00.02 | Marquis42 | Oh right, derp. |
| 16:00.05 | blitzrage | well... set it on fire anyways :) |
| 16:00.07 | Marquis42 | I forgot about that War of 1812. :) |
| 16:00.16 | blitzrage | it used to be pink marble, but it's white now because they had to paint it from the fire damage |
| 16:00.41 | blitzrage | something about the pink house doesn't work as well as the white house |
| 16:00.49 | Marquis42 | Indeed |
| 16:00.50 | blitzrage | so you're welcome! |
| 16:00.53 | Marquis42 | lol |
| 16:04.38 | blitzrage | went to war, lots of people died, borders stayed exactly the same |
| 16:05.18 | jsmith | blitzrage: Actually not marble -- but pink Aquia sandstone from Government Island (about five miles from my house) |
| 16:05.46 | jsmith | blitzrage: But then, it's all painted white today, so who cares, right? |
| 16:09.47 | pabelanger | In 1812 Canada technically didn't exist. |
| 16:09.56 | pabelanger | but we did still burn down the whitehouse |
| 16:09.57 | pabelanger | .trolo |
| 16:11.11 | pabelanger | Lower Canada I guess |
| 16:11.15 | pabelanger | was the right name |
| 16:11.19 | pabelanger | googling it now |
| 16:11.27 | Stefan26 | do you Digium-people have tools for stress-testing asterisk? |
| 16:11.54 | pabelanger | SIPp? |
| 16:12.10 | Stefan26 | Ive heard of SIPp |
| 16:12.22 | Stefan26 | a colleague used to try SIPp |
| 16:12.33 | pabelanger | kinda the defacto tool for stress testing SIP |
| 16:13.06 | Stefan26 | Can it test encryption and DTLS? |
| 16:13.17 | pabelanger | not sure about dtls |
| 16:13.26 | pabelanger | but if compiled with SSL, you can do tls |
| 16:13.42 | pabelanger | You also need to add flags for media |
| 16:13.54 | pabelanger | SIPp is more for call control, then media |
| 16:16.43 | Stefan26 | we built some load generating robots using pjsua library and some other testing robots for webrtc with SIPml5 and javascriipt, never bothered to learn SIPp yet |
| 16:17.31 | Stefan26 | i guess robots will never do as accurate testing as actual production systems will |
| 16:18.08 | pabelanger | Same, I'm working on something using pjproject for testing too |
| 16:18.21 | pabelanger | load isn't my goal, more external control via python |
| 16:18.28 | pabelanger | but would be cool to see what you did |
| 16:19.17 | Stefan26 | i can post em here but im at home they are at office |
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| 16:19.29 | Stefan26 | very little code and just 3 load profiles |
| 16:21.29 | Stefan26 | the problem is just that while we can get almost arbitrarily many pjprojects robots started at the same time, it's hard to get more than 10 webrtc robots at the same time |
| 16:29.25 | Marquis42 | What's the limiting factor on the webrtc robots? |
| 16:30.29 | pabelanger | Ya, was going to ask that |
| 16:34.14 | Stefan26 | that we havent built or seen a testing framework for it, each test-client is just javascript-code contained in a chrome browser |
| 16:34.44 | Stefan26 | limited to 4 per machine atm using all browsers based on chromium |
| 16:36.53 | Stefan26 | so at work we run 10-20 on different virtual machines but it's tedious starting and stopping all, would like to be able to start and configure all from one point like with sipp or the pjsua-robots |
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| 16:43.42 | Stefan26 | at least firefox seems to have the problem of a firefox-tab in the background cannot get getUserMedia() success until you focus it, Im not sure about chrome anymore |
| 16:44.54 | Marquis42 | wonders if there's some way to make it work in a Docker container, slightly trolls pabelanger in the process |
| 16:45.56 | Stefan26 | I'm sure there are many ways to build a test framework but it requires development effort |
| 16:46.58 | Marquis42 | Oh for sure, but Docker *might* be a solution to only getting a few instances spun up on a given machine (more than 4 per machine anyway). Then you "only" have the job of stitching it together with orchestration and such. |
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| 16:49.37 | blitzrage | pabelanger, you mean upper canada |
| 16:49.51 | blitzrage | Ontario was Upper Canada, and Quebec was Lower Canada |
| 16:50.33 | blitzrage | it's backwards because they landed on the east coast, so you have to think of it as a migration for lower vs upper (Quebec -> Ontario was the migration path, so Quebec was lower than Ontario....) |
| 16:50.51 | Stefan26 | im not even sure how popular webrtc will be in 5 years |
| 16:50.59 | Stefan26 | a lot of people seem to use softphones but not web based ones |
| 16:51.02 | blitzrage | Stefan26, I like Sippy Cup which makes building SIPp scenarios quite easy |
| 16:51.24 | blitzrage | Stefan26, I think getting webrtc == softphone out of your head is the first step :) |
| 16:51.43 | Stefan26 | ? |
| 16:51.52 | Stefan26 | it's a special kind of softphone |
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| 16:52.52 | blitzrage | well webrtc is more media in general, rather than just a softphone |
| 16:53.01 | blitzrage | although you can certainly build a softphone with webrtc tech :) |
| 16:53.27 | blitzrage | we'll be adding that this year if I can get the base platform built to support such a thing. Call centre in a web page essentially |
| 16:53.28 | Marquis42 | Exactly, it's definitely more than just traditional voice or voice/video. |
| 16:55.15 | Stefan26 | right |
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| 17:27.14 | skrusty | just passed the 10k mark of calls handled with ARI in production :o |
| 17:27.37 | skrusty | small fry i am sure, but still nice |
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| 17:36.11 | file | skrusty, still cool to know :D |
| 17:36.23 | file | skrusty, mind if I tweet it? |
| 17:38.18 | DivideBy0 | skrusty: in house calling, or a public app? |
| 17:38.27 | DivideBy0 | btw, congrats |
| 17:38.40 | file | or you can and I shall RETWEET |
| 18:04.40 | blitzrage | skrusty, nice moves! |
| 18:04.48 | blitzrage | skrusty, using any sort of proxy for that? :) |
| 18:09.30 | DivideBy0 | skrusty: how do you handle busy signals? I have to wait for an originated call to timeout since it never enters the ari |
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| 19:35.15 | skrusty | file: not at all :) |
| 19:35.52 | skrusty | file: which ever you'd prefer :) |
| 19:36.16 | file | I like to retweet :D |
| 19:36.21 | skrusty | hehe ;)( |
| 19:36.29 | skrusty | ;) even - stupid kb :/ |
| 19:38.20 | file | keyboards are great! they allow us to type |
| 19:39.56 | skrusty | blitzrage: as yet, no, but very soon i will be testing some ari proxying |
| 19:40.15 | blitzrage | nice :) |
| 19:40.34 | skrusty | DivideBy0: as it stands, not very well :) |
| 19:41.06 | skrusty | it's on my "to review" list |
| 19:41.50 | skrusty | my dial code is pretty big already - i am in the process of refactoring it :) |
| 19:42.06 | coreyfarrell | Stefan26: see tests/channels/SIP/sip_tls_register in the asterisk testsuite for an example of SIPp encryption |
| 19:42.26 | coreyfarrell | oops window was scrolled way up :( |
| 19:44.38 | Stefan26 | thanks i just got back from drinking beers |
| 19:45.33 | Stefan26 | That's TLS only, not DTLS? |
| 19:46.01 | Stefan26 | can browse through asterisk testsuite tomorrow at work |
| 19:46.49 | coreyfarrell | pretty sure that test is TLS.. not sure if SIPp does DTLS |
| 19:49.40 | Stefan26 | that's a shame, my goal is to find a good patch to asterisk's dtls-implenation which makes asterisk behave well with 50+ simultaneous peers using dtls (webrtc clients almost exclusively) but it's hard to make a test framework |
| 19:50.32 | skrusty | DivideBy0: my main issue with ARI now is session progress... or lack of it :) |
| 19:50.40 | Stefan26 | the jira is https://issues.asterisk.org/jira/browse/ASTERISK-24832, but our test only includes 10-20 peers at most now |
| 19:50.52 | Stefan26 | in production it crashed a lot and they intend to use 50 i think |
| 19:52.03 | Stefan26 | although 50 peers in production might be less stress than 10 aggressive calling robots |
| 19:55.55 | DivideBy0 | skrusty: my idea is to use the AMI to add that info. I just haven't tried yet. |
| 19:56.51 | skrusty | that would work, but i think there needs to be a new channel stateof some sort |
| 19:58.33 | DivideBy0 | yeah, we're subscribed to the channel's events, so maybe just have the ari emit more session events to us? |
| 19:59.38 | skrusty | well, there's rinigng/up/down etc, what's needed, i think, is a progress state |
| 20:00.36 | skrusty | as for busy, i've not looked into this properly yet,ami right in thinking you just get a channel destroyed event? |
| 20:02.13 | coreyfarrell | Stefan26: sorry I don't know much about TLS/DTLS, I just knew of a SIPp test that might be in the right direction for you to look at |
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| 20:04.23 | skrusty | DivideBy0: does the channel destoryed event not contain the reason? |
| 20:05.06 | skrusty | Cause (int) and Cause_txt |
| 20:05.34 | skrusty | DivideBy0: cause_txt - Text representation of the cause of the hangup |
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| 20:14.13 | DivideBy0 | I've looked at those. lemme look again. but I have to brb |
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| 20:24.01 | skrusty | DivideBy0: let me know! |
| 20:24.06 | DivideBy0 | looking right now |
| 20:24.16 | skrusty | cool :) |
| 20:33.04 | DivideBy0 | turns out it's not busy, it's the fast busy I'm having trouble with. I'm using the causes just like you said |
| 20:34.33 | skrusty | ah cool |
| 20:34.38 | DivideBy0 | now that I'm playing with it, it seems like it's the new * system I'm using for the ARI, and not the ARI itself |
| 20:34.59 | skrusty | tbh, im still using 12.7 in production atm |
| 20:35.27 | DivideBy0 | I wasn't planning on jumping on that problem today, but thanks. at least I have something checked. I'm using 13.1/ stock freepbx |
| 20:35.42 | skrusty | ah |
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| 21:17.00 | gtjoseph | can someone try compiling app_voicemail in trunk with IMAP_STORAGE set? |
| 21:28.26 | mjordan | skrusty: if you had the concept of an "early bridge" - that is, where two channels could be placed into a bridge and progress indications from an outbound channel are relayed to the other channel - would you still need progress events? |
| 21:30.48 | gtjoseph | wdoekes: a few months ago in commit 426706 you put an "ast_assert(msgnum < vms->msg_array_max);" in app_voicemail.c around line 2444. It doesn't compile because neither msgnum nor vms exist. |
| 21:31.03 | gtjoseph | Can I remove it? |
| 21:31.18 | gtjoseph | or does it serve some purpose? |
| 21:35.00 | mjordan | was worried it was him |
| 21:35.07 | mjordan | not wdoekes, myself. |
| 21:35.12 | mjordan | bah. |
| 21:35.28 | gtjoseph | oops, looked like his commit |
| 21:40.00 | gtjoseph | mjordan: looks like the issue goes back to 11. Shall I just fix it? |
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| 21:52.28 | mjordan | gtjoseph: sure! That'd be great |
| 21:52.45 | gtjoseph | will do. Is 12 still alive or is it EOLd? |
| 21:53.09 | gtjoseph | make clean |
| 21:56.07 | mjordan | EOLd |
| 21:56.14 | mjordan | well, security fix only |
| 21:56.22 | gtjoseph | cool. I'm just going to remove the assert. |
| 21:56.28 | gtjoseph | from 11, 13, trunk |
| 21:56.30 | mjordan | ack |
| 21:56.34 | mjordan | sounds good to me |
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| 22:53.05 | rmudgett | file: There doesn't seem to be a good way to create a default sorcery object if no objects are configured of that type. i.e., If pjsip.conf type=global then how can a default object be created if it doesn't exist in the conf file. |
| 23:01.01 | file | you can't create a default object and have it be stored in sorcery, you have to do it outside |
| 23:01.29 | file | how it's used really dictates how that should be done |
| 23:02.06 | file | other types create one and apply it if it does not exist, with the act of applying setting stuff |
| 23:03.40 | rmudgett | The only one that seems to create a default object for config is the pjsip.conf type=system. That is not reloadable. |
| 23:07.05 | gtjoseph | if you needed to, couldn't you add a memory wizard for the object (before the config file one), then add 1 instance to it? |
| 23:07.22 | file | just because you can, doesn't mean you should |
| 23:07.49 | gtjoseph | just tossing out an idea |
| 23:08.24 | rmudgett | There should only be zero or one global type object in pjsip.conf |
| 23:08.31 | file | so - what do you need? |
| 23:09.18 | rmudgett | So right now if there is no global type object the defaults are not applied and thus the Max-Forwards header does not get put on SIP messages. |
| 23:09.39 | rmudgett | https://issues.asterisk.org/jira/browse/ASTERISK-24807 |
| 23:09.41 | file | correct. |
| 23:11.24 | rmudgett | I'm looking for a place to determine if the defaults need to be applied if there is no global object. It's looking like I need to create a new function that the initial load and the reload needs to call to check if there was any global object. |
| 23:12.15 | rmudgett | Also it looks like if there is more than one global type object the last one will be used. |
| 23:12.18 | file | add an observer for the object type which is invoked on loaded, do a sorcery retrieve on it and if nothing is returned allocate a global type and apply it |
| 23:12.45 | rmudgett | Ah that might work better. |
| 23:13.46 | file | the other stuff configured by that type already take lack of it into account, so it really only matters for the headers |
| 23:18.58 | rmudgett | Well default_outbound_endpoint doesn't return the default. It acts as if the default is not defined or empty. |
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