| 00:09.03 | jameswf | malcolmd can you (or anyone I suppose) take a gander at: http://issues.freepbx.org/browse/FREEPBX-7947 . It looks like something weird with chan_dahdi. Our build and the piaf build don't seem to like the us tonezone. |
| 00:09.34 | jameswf | See if it is worth pointing them to your bug tracker |
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| 10:49.51 | josemslopes | i am trying to play holdtime when some caller on callqueue press 0, so i decided right now to develop in the app_queue.c. (http://pastebin.com/pg2LzunH) |
| 10:50.11 | josemslopes | i am a newbie on developing on asterisk, so if anyone can guide me on this development, i will thank you. |
| 10:50.27 | josemslopes | the development it is working, but right now i am facing a problem, when a caller press 0 to listen holdtime, and the call is attended by a agent ... the agent only ear the caller, when the playing of holdtime ends. |
| 10:50.51 | josemslopes | it is possible to check when the agent answer the call from the queue for a queue entry, so i can skip the playing of holdtime ? |
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| 13:13.57 | anonymouz666 | oej: tjenare! |
| 13:14.21 | anonymouz666 | oej: is any of your branches has worked with LYNC 2013 with SIP/TLS and SRTP? I have a scenario that TLS is working OK, but lync's SRTP can't be decoded by Asterisk. Already see this before? |
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| 13:51.12 | malcolmd | jameswf: they're a card customer. for what they're reporting, they should contact our support team directly and work with them to troubleshoot and triage. i dropped a note on FREEPBX-7947 to that affect. |
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| 13:58.47 | jameswf | malcolmd: sounds good to me... If you find something I should make our folks aware of let me know. |
| 13:59.04 | jameswf | assuming they contact you |
| 13:59.10 | jameswf | *yall |
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| 14:25.17 | malcolmd | jameswf: yop. i don't have a rig to do any personal testing, so i'll have to leave it to the support guys to troubleshoot and triage |
| 14:28.06 | oej | anonymous666: Th teapot branch has a few srtp fixes. |
| 14:28.13 | oej | As well as other Lync stuff |
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| 14:54.56 | anonymouz666 | the teapot branch is automatically in sync with the latest 1.8 code? |
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| 14:57.51 | anonymouz666 | I will need to diff the res_srtp, chan_sip, sdp_crypto to see the specifics fixes for the SRTP code, in fact, I want to port to Ast 11. |
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| 15:19.02 | mjordan | jrose_atDigium: I'm not sure I understand some of your comments on r3990 |
| 15:19.05 | mjordan | what do you mean by: |
| 15:19.13 | mjordan | "Honestly, I'm not sure this is entirely adequate, but it seems to work in all of the conditions I've tried so far and it doesn't impede normal attended transfers. I might need to try and figure out what happens if I masquerade in a channel that is already dialing though. I'm not sure if that's something we can really expect to happen under normal conditions, but that seems like something that would screw up this approach." |
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| 15:26.18 | jrose_atDigium | mjordan: I'm not sure if there are any conditions where something can masquerade into a channel that is in the middle of a dial where this is the case, but what if a channel that masqueraded into another dialing channel was already dialing? |
| 15:27.51 | mjordan | jrose_atDigium: for that to happen, the channel that was masqueraded out (who was dialling) would have to survive the operation. Generally, that isn't the case. |
| 15:27.56 | mjordan | Redirect won't do that. |
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| 15:28.14 | mjordan | Generally, you can't initiate a call pickup if you're already dialling someone |
| 15:28.24 | mjordan | And you generally can't transfer unless you're in a bridge. |
| 15:28.32 | jrose_atDigium | In that case, it's probably fine. |
| 15:28.33 | mjordan | Also: why did you make the patch only in the dial API? |
| 15:28.45 | mjordan | I would suspect that app_dial would have this problem |
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| 15:29.35 | jrose_atDigium | mjordan: Actually the dial API just contains the API commands for handling it. The invocation of it all occurs in stasis_channels |
| 15:30.08 | mjordan | Yeah... |
| 15:30.19 | mjordan | I'll have to think about that. I'm not sure that should be done in Stasis channels either. |
| 15:30.44 | mjordan | I'll think about it. Having this be a public API in the dial API feels wrong, but I'll see what other options we have. |
| 15:30.53 | jrose_atDigium | alright |
| 15:35.44 | jrose_atDigium | mjordan: Actually, as is I don't think there is anything dial API specific, so all of that stuff could be moved into stasis_channels and invoked statically. No need for a public API there. If we decide that setting all of this up off of the ast_channel_publish_dial function isn't appropriate, then I don't know. |
| 15:36.51 | mjordan | I'm not sure it should be in stasis channels either. |
| 15:37.18 | mjordan | There's some risk there, as the thing that "knows" what is going on with the channels in question is the actual thing controlling the dial operation |
| 15:37.26 | mjordan | I'll think about it some more. |
| 15:40.55 | seanbright | so there is this variable called MONITOR_FILENAME in app_queue that you can set to control the recording's filename |
| 15:41.47 | seanbright | there's also a function - escape_and_substitute - that converts ^{FOO} to ${FOO} so you actually set MONITOR_FILENAME from dialplan |
| 15:42.00 | mjordan | oh good lord app_queue has monitor built into it |
| 15:42.37 | seanbright | but something else that escape_and_substitute does is replace a , with \, |
| 15:42.40 | Marquis42 | Holy jeez, really? It's even worse than I thought... |
| 15:43.41 | mjordan | seanbright: when do you put a ',' in the filename? |
| 15:43.42 | seanbright | and i have no idea why it does that |
| 15:44.00 | leifmadsen | o.O |
| 15:44.02 | leifmadsen | wow. |
| 15:44.17 | seanbright | mjordan: Set(MONITOR_FILENAME=Something-^{CUT(MEMBERINTERFACE,_,2)}-... |
| 15:44.20 | seanbright | as an example |
| 15:45.40 | seanbright | i can't think of any reason to escape commas |
| 15:46.10 | mjordan | I'm not sure why it substitutes the ',' either. I looked at the svn history, and that was part of the 'holy crap we broke all of app_queue' in 12 work |
| 15:46.13 | seanbright | sorry... did i miss a joke? why is monitor being built in to app_queue a big deal? |
| 15:46.38 | seanbright | mjordan: http://svnview.digium.com/svn/asterisk?view=revision&revision=83484 |
| 15:47.25 | mjordan | does not look forward to the day when Monitor dies |
| 15:47.51 | mjordan | I mean, I do look forward to it, I do not look forward to the subtle ways in which Monitor in Queue is different than Monitor in dialplan (or the weird crap that happens in channel.c) |
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| 15:47.58 | mjordan | but! Anyway. |
| 15:48.10 | mjordan | putnopvut: I think you may have been involved in the code in question |
| 15:48.29 | putnopvut | reads back |
| 15:48.34 | seanbright | but even if i did: |
| 15:48.35 | mjordan | Do you know why we might replace ',' with '\' when parsing out embedded variables in MONITOR_FILENAME? |
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| 15:48.47 | seanbright | mjordan: it's replaced with \, |
| 15:48.55 | mjordan | ah, we escape the comma |
| 15:48.58 | seanbright | yeah |
| 15:49.24 | putnopvut | Um...no. As strange as it may sound, that code is before my time. Let me put my thinking cap on. |
| 15:49.31 | putnopvut | chugs whiskey |
| 15:49.46 | putnopvut | Oh! |
| 15:49.47 | putnopvut | I bet I know |
| 15:49.54 | seanbright | i think it's backwards |
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| 15:50.08 | seanbright | if you do SET(MONITOR_FILENAME=Foo,Bar.sln) |
| 15:50.17 | seanbright | then that would be interpreted as two arguments to SET, correct? |
| 15:50.22 | putnopvut | I think the Monitor() application is invoked directly, and so the idea there is to not make the Monitor() application parse the file name as multiple options to Monitor() |
| 15:50.31 | seanbright | ahhhh |
| 15:50.51 | seanbright | then the comma should be scaped _after_ variable substitution |
| 15:50.53 | putnopvut | That's the best I can come up with |
| 15:50.55 | seanbright | escaped* |
| 15:50.57 | putnopvut | Yeah |
| 15:51.12 | putnopvut | app_queue: not even once |
| 15:51.18 | seanbright | yeah, well |
| 15:51.28 | seanbright | is there a replacement? |
| 15:51.29 | seanbright | :D |
| 15:51.45 | file | I dunno, have you written one yet? :P |
| 15:52.22 | seanbright | heh |
| 15:52.36 | seanbright | yes... but it's written in java |
| 15:52.39 | seanbright | will merge |
| 15:57.57 | seanbright | so fixing this would be a breaking change |
| 16:01.27 | mjordan | seanbright: does the current behavior of escaping the commas work in any useful scenario? |
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| 16:04.12 | seanbright | i can't think of one |
| 16:04.45 | mjordan | I'm willing to go with "bug fix" |
| 16:04.58 | file | I have a bad feeling. |
| 16:05.06 | mjordan | file: because app_queue? |
| 16:05.31 | file | oh it's just isolated there |
| 16:05.32 | file | nevermind! |
| 16:05.42 | file | bad feeling gone |
| 16:05.46 | mjordan | probably should make a quick issue in JIRA for it, just so it gets documented when we break someone's work around |
| 16:09.59 | seanbright | ok. i will get around to it in a bit. locally i just removed that case statement and it works. but it's not ideal. |
| 16:10.24 | seanbright | if we wait until october 21st that would be even better |
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| 16:59.11 | putnopvut | In ari-py, I can register callbacks to be called when an event occurs. Is there a way for me to then tell ari-py I am no longer interested in that event? Or a way to tell ari-py that I only want to be informed of this event one time? |
| 16:59.38 | putnopvut | Looks like the node-js library has a .once function that can be used to indicate you only want to be called into one time. |
| 16:59.46 | putnopvut | Not seeing anything similar in ari-py though. |
| 17:00.57 | file | yeah, I didn't see anything when I was writing something |
| 17:01.49 | putnopvut | Hm, I see an embedded EventUnsubscriber class in the on_event callback in the client. |
| 17:02.04 | putnopvut | I suppose that when my event is called into, there's some mechanism I can use to unsubscribe myself at that point. |
| 17:02.25 | mjordan | putnopvut: yes |
| 17:02.31 | putnopvut | Ah it's the return of on_event. |
| 17:02.34 | putnopvut | I see how to do it now. |
| 17:02.42 | mjordan | when you register the event, you get back the unsubscriber and yes I am now talking to the past |
| 17:02.48 | putnopvut | heh |
| 17:02.56 | putnopvut | yay for text chats |
| 17:03.05 | file | yay past |
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| 17:26.15 | mjordan | file: I can't recall; what was the definition of the proxy_media attribute used during bridge creation? |
| 17:26.40 | file | forces media to be proxied through Asterisk |
| 17:27.42 | mjordan | right, but how does that end up being different than dtmf_events? |
| 17:28.02 | file | if DTMF is carried in a signaling manner then media does not need to be proxied |
| 17:28.26 | mjordan | true, but I don't really see a specific bridge flag being set for that |
| 17:29.44 | file | hrm, I may have done it just in case we do make the differentiation one day if ever |
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| 17:36.34 | anonymouz666 | is building Asterisk 11 heavily patched with oej's rtcp/srtp improvements. |
| 17:37.19 | anonymouz666 | the call quality realtime stuff is very nice |
| 17:37.57 | mjordan | anonymouz666: A good chunk of pinefrog actually made it into 12. The reporting mechanisms didn't, but would be pretty easy to add as a separate module. |
| 17:38.09 | mjordan | That's pretty much how res_hep_rtcp works. |
| 17:38.28 | anonymouz666 | hep is the same protocol that sip homer uses? |
| 17:38.41 | mjordan | yup |
| 17:39.10 | mjordan | Homer supports sending it a variety of things |
| 17:39.28 | mjordan | Asterisk can send both RTCP stats and SIP traffic (PJSIP stack only) |
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| 17:42.35 | file | PJSIP is hawt |
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| 17:56.52 | jsmith | Yay, my pjproject package for Fedora/RHEL/CentOS got accepted |
| 17:57.12 | jsmith | Now building a bunch of RPMs in the Fedora build system |
| 18:01.54 | pabelanger | jsmith, beers to be provided |
| 18:03.04 | jsmith | pabelanger: You keep forgetting, I don't drink |
| 18:03.13 | Marquis42 | Hookers then? ;) |
| 18:03.15 | Marquis42 | is kidding |
| 18:03.33 | Marquis42 | Nice moves, though! Glad to see that hurdle jumped. |
| 18:03.46 | pabelanger | jsmith, do I keep forgetting? |
| 18:03.47 | pabelanger | do I? |
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| 18:17.29 | mjordan | jsmith: That's awesome. Thanks for doing all the hard work and getting that in |
| 18:17.51 | jsmith | mjordan: Well, a lot of that hard work was just being patient for some fixes in pjproject itself |
| 18:18.04 | jsmith | mjordan: All in all, it was only a few hours of work on my side |
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| 18:25.53 | pabelanger | file, about to test unicastrtp, would this be a dial string? Dial(UnicastRTP/127.0.0.1:1234/res_rtp_asterisk/ulaw) |
| 18:25.58 | pabelanger | not sure what engine should be |
| 18:26.03 | file | sure, or leave engine empty |
| 18:26.10 | pabelanger | ok |
| 18:26.12 | file | (I have an example on my blog post) |
| 18:26.15 | pabelanger | ok |
| 18:26.20 | file | (including ffmpeg part) |
| 18:27.58 | jsmith | pabelanger: Have each of your agent bridges with a UnicastRTP channel, so that supervisors can simply monitor calls by listening to different internet radio stations :-p |
| 18:28.53 | pabelanger | jsmith, indeed |
| 18:29.25 | pabelanger | actually, I was trying to figure out a way to have asterisk use ffplay so I could stream icecast streams into asterisk would out asterisk doing transcoding |
| 18:29.45 | pabelanger | the concept was to modify chan_multicast_rtp to do unicast |
| 18:30.04 | pabelanger | however, the customer didn't want to sponsor the work effort for it |
| 18:30.10 | pabelanger | and went a different route |
| 18:30.19 | jsmith | Understood |
| 18:30.42 | pabelanger | but looks like file picked up and did the work, so going to give it a test with my POC I was doing for this client |
| 18:30.59 | file | not with a focus on that so if it is not achievable don't be surprised |
| 18:31.22 | pabelanger | I actually think is should work out of box |
| 18:31.28 | pabelanger | with some minor dialplan updates |
| 18:33.11 | mjordan | Hm. I think audiohooks should probably always set the UNBRIDGE soft hangup flag when they are attached to a channel. |
| 18:34.12 | mjordan | ponders that change some more |
| 18:34.57 | file | mjordan, audiohooks yes |
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| 19:32.47 | mjordan | yo dawg. I heard you want to bridge your bridge with bridges. |
| 19:33.00 | sgalarneau | recursive much? |
| 19:33.19 | sgalarneau | so I'm seeing some strange behavior with trunk |
| 19:33.25 | mjordan | so... do you know what path the swagger client is using to play on the bridge? |
| 19:33.42 | sgalarneau | during first test subscribing to bridge events did nothing, and now unsubscribing doesn't change the list of bridges subscribed to |
| 19:33.51 | sgalarneau | mjordan: 1 sec |
| 19:34.24 | sgalarneau | ari/bridges/1/play?media=sound%3A... |
| 19:34.48 | mjordan | hm. |
| 19:34.56 | mjordan | I'm confused why this work in Python. |
| 19:35.00 | mjordan | works. |
| 19:35.08 | sgalarneau | I bridged a physicial phone with a softphone, reversing the order of the originates/bridgeAddChannel and in both cases, the soft phone receives playbacks but the physicial phone does not |
| 19:35.27 | sgalarneau | physical phone could hear playback when it was the only channel in the bridge |
| 19:35.31 | mjordan | what do you get back from ARI? Or what does the Asterisk CLI show? |
| 19:35.42 | sgalarneau | even after soft phone leaves, physical phone can no longer get any playback through bridge |
| 19:36.02 | sgalarneau | ARI is returning 200/204s as expected, which is weird |
| 19:36.13 | sgalarneau | I can even see the events that playback started/finished |
| 19:36.48 | mjordan | media problems then. |
| 19:36.51 | mjordan | Does it happen in 12? |
| 19:37.09 | sgalarneau | for semidy yes, haven't tried it myself |
| 19:37.14 | mjordan | I'd try 12. |
| 19:37.30 | mjordan | If media is the problem, I would not expect that to occur in 12. |
| 19:37.47 | mjordan | but at the same time, this could also be something else. |
| 19:41.42 | sgalarneau | I see a warning about a codec mismatch for the physical phone but that's it |
| 19:41.47 | sgalarneau | need to test with 12 next |
| 19:42.11 | mjordan | welp, that would do it |
| 19:42.37 | sgalarneau | "setting write format to slin16 from g722 native formats" |
| 19:43.21 | sgalarneau | I'll be back in a bit when 12 is compiled and running |
| 19:44.10 | mjordan | Hm. |
| 19:44.28 | mjordan | I think we'd have to look at (a) a sip trace of what the physical phone offered/answered with, and (b) what the endpoint was configured with |
| 19:44.40 | mjordan | we do know of one bug currently that richard found, so you may be running into that |
| 19:45.01 | sgalarneau | I'm assuming you are using pjsip? |
| 19:45.07 | mjordan | yes |
| 19:45.23 | sgalarneau | that phone is using sip |
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| 19:45.36 | mjordan | well, no idea bout that then |
| 19:51.11 | mjordan | sgalarneau: so... yeah. Either there's a bug in the javascript swagger client, or in ari-client, as doing a /play on a bridge through ari-py works just fine |
| 19:52.45 | sgalarneau | with multiple parties in a bridge? |
| 19:53.01 | sgalarneau | works fine with 1 channel in a bridge for me with both the swagger client and ari-client |
| 19:54.28 | mjordan | nope, holding bridge, one party |
| 19:55.07 | mjordan | HM. |
| 19:55.12 | mjordan | This may be something in my code. |
| 19:55.14 | mjordan | looks some more |
| 19:56.41 | sgalarneau | we're using a mixing bridge, based on scott and ben's code |
| 19:57.37 | mjordan | Different app |
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| 19:57.56 | mjordan | doh. Used 'playback.once' instead of 'playback.on' |
| 19:58.03 | mjordan | works now. |
| 19:58.15 | mjordan | huzzah for that at least. |
| 19:58.50 | sgalarneau | it happened once to me :) |
| 20:00.08 | mjordan | goes off to add node.js to his resume |
| 20:00.54 | sgalarneau | you can do that as soon as you've used npm install |
| 20:01.00 | mjordan | ooo |
| 20:01.07 | mjordan | I must be an expert now then |
| 20:01.14 | file | does that mean if you use cpan you can add perl to your resume? |
| 20:01.32 | sgalarneau | no, no one can read their own perl code, so that doesn't count |
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| 20:01.36 | mjordan | file: who would want to admit to that though |
| 20:02.02 | file | the act of not admitting is an admission in and of itself |
| 20:02.14 | mjordan | file: I refuse to admit that |
| 20:02.36 | opticron | so you're saying resumes should have a Confessions section? |
| 20:02.44 | sgalarneau | there are a few languages I prefer not to list on my resume |
| 20:08.53 | sgalarneau | same behavior with 12 |
| 20:08.57 | sgalarneau | :( |
| 20:09.56 | sgalarneau | and also not subscribing to bridge events, I get a 200 but the bridge_ids array is empty |
| 20:14.31 | sgalarneau | I've got a debug log with debug level set to 5 |
| 20:19.53 | mjordan | send it to me |
| 20:20.58 | sgalarneau | done |
| 20:23.37 | mjordan | not that this shouldn't work, but what is the config for sipphone-00000001? |
| 20:23.40 | mjordan | or rather, sipphone? |
| 20:24.49 | mjordan | sgalarneau: pretty sure you shouldn't be URI encoding the ':' in bridge:1 |
| 20:25.14 | sgalarneau | that would be the swagger-client doing it |
| 20:25.23 | mjordan | that's not kind |
| 20:25.24 | sgalarneau | I've gotten it to work, it's just not consistent |
| 20:25.43 | sgalarneau | subcribe/unsubscribe that is, sometimes works |
| 20:25.44 | mjordan | hm |
| 20:25.47 | mjordan | that being said... |
| 20:25.54 | mjordan | [Sep 12 15:09:12] DEBUG[2132] stasis/app.c: Bridge '1' is 7 interested in test |
| 20:25.59 | mjordan | so it did bump the subscription to it |
| 20:26.30 | mjordan | I think the response JSON is just wrong. You are subscribed to the bridge. |
| 20:27.04 | sgalarneau | sent you the sip.conf config for sipphone |
| 20:29.47 | mjordan | sgalarneau: file an issue for the app subscription thing. You are getting subscribed, but the app model isn't getting updated with the event sources correctly |
| 20:30.14 | sgalarneau | will do |
| 20:30.23 | sgalarneau | I'll include the debug log |
| 20:32.54 | mjordan | so, there's something very strange going on with that SIP channel. |
| 20:33.03 | mjordan | try pjsip first and see if that fixes it. |
| 20:33.16 | mjordan | That aside, I'd file an issue with your sip.conf and the codec mismatch |
| 20:36.15 | sgalarneau | will do |
| 20:38.54 | sgalarneau | weird thing is that sip config is a template, softphone uses the same config, wonder why it's working |
| 20:39.32 | file | jrose_atDigium, re: ASTERISK-24290 it's because it resolves the value given so if it's a host it can get all addresses |
| 20:42.12 | jrose_atDigium | file: I'm not sure I know what more than half of those 'it's are just yet. |
| 20:42.39 | jrose_atDigium | Mind rephrasing that in a less pronoun heavy way? |
| 20:47.26 | file | the code takes the value as provided and attempts to resolve it as if it were a hostname |
| 20:47.35 | file | in the case of an IP address this doesn't resolve and the IP address is returned |
| 20:47.42 | file | in the case of a hostname all addresses are returned |
| 20:47.50 | file | since adding a netmask makes the value invalid this fails |
| 20:48.11 | jrose_atDigium | alright, thanks |
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| 21:28.06 | mjordan | sgalarneau: threw some feedback on your dial example.js |
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