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| 08:24.53 | Elio19 | Is there an easy easy way to get opus and dtls sha-256 fingerprint support ? The Opus patch is for version 11.1.12, and sha-256 was not added until after 11.8 or so |
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| 09:33.48 | babak | WIMPy: Hi |
| 09:34.33 | babak | if a call was already forwarded on carrier network, and forwarded call comes to your server by PRI link , you can see 3 numbers ? A_number,B_number , original called number ? |
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| 13:09.47 | nuclear_horse | hi all. i have migrated from 1.8 with chan_sip to 13.2 with chan_pjsip and happy with it, but i have some issues |
| 13:11.13 | nuclear_horse | when i call multiple endpoints (PJSIP/100&PJSIP/101 for example), 100 answer the call, then 101 have new missed call |
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| 13:11.31 | nuclear_horse | 'c' option of bridge doesn't help anymore |
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| 13:11.43 | nuclear_horse | of dial |
| 13:12.06 | nuclear_horse | can somebody help me? |
| 13:12.12 | Elio19 | Hi |
| 13:13.00 | [TK]D-Fender | nuclear_horse:Show us a call with pjsip deebug enabled |
| 13:13.52 | nuclear_horse | ok, i'll try |
| 13:14.59 | Elio19 | Hey [TK]D-Fender, do you know how ppl are going about WebRTC these days? I tried patching in opus but it did not go smooth |
| 13:15.19 | Elio19 | I was even editing .c files due to a bug ( sha-2 vs sha-256 ) |
| 13:15.44 | [TK]D-Fender | Elio19: Never touched it |
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| 13:16.54 | Elio19 | Auth and stuff works wss -> tls but as soon as asterisk starts to play a sound the webrtc client dies with format error :-/ |
| 13:17.20 | Elio19 | How can i find what format Asterisk is sending to the client? |
| 13:24.28 | *** join/#asterisk snmp (~q@95.47.164.1) |
| 13:24.43 | snmp | hola! |
| 13:25.13 | nuclear_horse | [TK]D-Fender, http://pastebin.com/G9iAdsTG is it ok? |
| 13:25.36 | snmp | asymmetric nat drives me mad |
| 13:26.05 | snmp | my brain is melting |
| 13:26.07 | snmp | if any |
| 13:27.04 | snmp | in case anyone cares |
| 13:27.14 | snmp | got two iax peers |
| 13:27.50 | snmp | first is handwritten with configs |
| 13:28.11 | snmp | second is a realtime based |
| 13:28.30 | snmp | everythings ok with first |
| 13:28.50 | snmp | registred port is 4569 and percieved port is 4569 |
| 13:29.00 | snmp | but the second one.. |
| 13:30.08 | snmp | =( |
| 13:31.34 | [TK]D-Fender | nuclear_horse: Three is no Dial with a "c" in tehre |
| 13:31.48 | [TK]D-Fender | needs to RMA his keyboard.... |
| 13:32.46 | nuclear_horse | [TK]D-Fender: in this case no, i removed it after it didnt help |
| 13:33.07 | nuclear_horse | i'll add it and try again |
| 13:36.07 | [TK]D-Fender | You shouldn't ask why it doesn't work and then show us a call where you specifically aren't even trying to do it. |
| 13:37.40 | Elio19 | Which versions are hot right now? |
| 13:38.18 | Elio19 | I've got about 5 different ones in my home folder but none of them are |
| 13:38.24 | Elio19 | ...really jumping |
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| 13:39.16 | [TK]D-Fender | latest 13 as listed in the topic |
| 13:39.40 | Elio19 | I've got 13.2 compiled and installed right now |
| 13:40.14 | Elio19 | its not doing the webrtc :-/ |
| 13:40.32 | Elio19 | (i did not try pjsip tho because i have a realtime sip peer thing) |
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| 13:42.46 | nuclear_horse | [TK]D-Fender: sorry, here the new one http://pastebin.com/hp55h6UN |
| 13:44.19 | [TK]D-Fender | [Mar 16 21:35:50] Reason: SIP;cause=200;text="Call completed elsewhere" |
| 13:44.23 | [TK]D-Fender | 467 |
| 13:44.43 | [TK]D-Fender | Sure looks like it's doing it |
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| 13:50.33 | nuclear_horse | [TK]D-Fender: i see, but missed call was added in list, and not added in 1.8 with chan_sip |
| 13:50.45 | Elio19 | Hey, would anyone like to help me with Asterisk? |
| 13:50.55 | [TK]D-Fender | ~ask |
| 13:50.56 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 13:50.57 | [TK]D-Fender | ^^^ |
| 13:51.17 | Elio19 | I'll take that as a, "yes." |
| 13:52.17 | Elio19 | So, its my goal to do a browser phone, ie WebRTC. And ive been working at it ceaselessly for quit some time. |
| 13:53.02 | Elio19 | I did compile with patch, i tried distro packages, even editing the .c files by hand to fix little bugs |
| 13:54.34 | Elio19 | After successfull compile and configure WS, WSS, TLS, DTLS, ICE, RTP, and apvt I'm at the point where i just want it to work |
| 13:55.19 | Elio19 | ...because i resolved so many dependencies and bugs my self |
| 13:55.42 | Elio19 | But now i dont even get an error any more, it just does not work |
| 13:57.01 | Elio19 | Client drops the call and complains about bad format "Bad Media Description" , which i think refers to "a=rtpmap:109 opus/48000/2" |
| 13:57.25 | Elio19 | but, "c=IN IP4 0.0.0.0" also looks suspicious |
| 13:59.27 | *** part/#asterisk y_ateya (~y_ateya@41.45.10.81) |
| 13:59.54 | [TK]D-Fender | Elio19: You're already modding the source. This means we can't tell what you've actually done here. You need to take this to #asterisk-deev |
| 14:00.05 | [TK]D-Fender | #asterisk-dev |
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| 14:00.57 | Elio19 | Well i just added an elseif so that Asterisk would recognize "sha-2" fingerprints as "sha-256" |
| 14:01.09 | Elio19 | Because FF sends the string "sha-2" vs "sha-256" |
| 14:01.19 | Elio19 | ok ill check that one out |
| 14:01.38 | [TK]D-Fender | Elio19: That contact IP is cleearly wrong though |
| 14:01.46 | Elio19 | btw, that is not needed in 13.2, that was for 11.2-11.11 |
| 14:01.48 | [TK]D-Fender | Elio19: Go prove your WAN IP is resolving, etc |
| 14:01.55 | Elio19 | Oh, its not |
| 14:02.04 | Elio19 | i had to use hosts file and i dont know why |
| 14:02.05 | [TK]D-Fender | Well then go look at that specifically |
| 14:02.30 | Elio19 | What system is that part of ICE? |
| 14:03.04 | Elio19 | I said that backwards: Is that contact field part of ICE? |
| 14:05.44 | [TK]D-Fender | no idea |
| 14:05.51 | [TK]D-Fender | I don't touch webrtc |
| 14:09.15 | file | the answer to that question would be no |
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| 14:09.44 | Elio19 | file, do you have some relevant information for me? |
| 14:09.50 | file | nope |
| 14:10.00 | Elio19 | Seems like you might... |
| 14:10.02 | file | I mean, the answer to your "Is that contact field part of ICE?" is no |
| 14:10.41 | Elio19 | Well, do you know where one might find the setting for that contact field? |
| 14:11.12 | Elio19 | But, im not sure if thats the right question because im pretty sure its a DNS issue |
| 14:11.30 | [TK]D-Fender | I'm pretty sure you could prove that in about 2 seconds... |
| 14:11.54 | file | the c= line doesn't matter in WebRTC |
| 14:12.02 | file | ICE and candidates are used to determine the flow for media |
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| 14:13.15 | Elio19 | ok, that make sense because the only error from the client is, "bad media" so i must be in contact to get any media at all... |
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| 14:28.48 | WIMPy | babak: Yes |
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| 15:56.42 | Elio19 | "c=IN IP4 0.0.0.0" relates to NAT |
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| 15:58.27 | file | ...? |
| 15:59.32 | [TK]D-Fender | Relates to "Asterisk can't lookup what you told it to to get it's WAN IP |
| 16:01.21 | mjordan | is pretty sure that's an old school way of putting a phone on hold |
| 16:01.31 | mjordan | but there's not a lot of context in that statement. |
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| 16:14.28 | Elio19 | No, thats what happens when you nat=comedia |
| 16:14.50 | Elio19 | ...think aboutit |
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| 16:35.11 | ChannelZ | mmmno |
| 16:38.12 | Elio19 | "Rejecting secure video stream without encryption detail" is an ambiguous error because only one of three conditions triggering that error are crypto related. |
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| 16:49.06 | Elio19 | When i got here i was wondering if i still had "it" with regard to open source software... |
| 16:52.00 | file | Elio19, what happens when you nat=comedia... |
| 16:53.17 | Elio19 | Asterisk sends the media right back where it got it from, so it does not need a c address |
| 16:53.34 | file | one side doesn't, yeah |
| 16:54.34 | Elio19 | file, |
| 16:54.46 | Elio19 | do you use webrtc? |
| 16:55.07 | file | no |
| 16:55.15 | Elio19 | pjsip? |
| 16:55.22 | file | yes |
| 16:56.32 | Elio19 | I spent so much time with out pjsip already its time for me to use pjsip. |
| 16:56.51 | [TK]D-Fender | That isn't a good basis for a decision |
| 16:57.05 | Elio19 | Its time based |
| 16:57.37 | Elio19 | IMO, it would be alot easier to read about it pjsip if the wiki browser ui didnt look like iframes from the 90s |
| 17:01.14 | WIMPy | Or some of them sometimes go missing. |
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| 17:15.28 | zamba | what is pjsip? i mean, is it a software? |
| 17:16.30 | zamba | it's a communication library? what does that mean? which software implements it? |
| 17:17.58 | [TK]D-Fender | Asterisk. |
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| 17:22.08 | Cuzner | [TK]D-Fender: done any localization for Colombia? :) |
| 17:25.00 | file | PJSIP is. |
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| 17:57.04 | newtonr | zamba, https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip |
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| 21:20.27 | pjensen00 | So I have an AGI application that takes a handoff from Asterisk on all incoming calls. I'm migrating to an ARI based system. What I do is spawn a new process that creates a Stasis app with the call ID generated from our SIP server. I'm trying to cut out the need to have the AGI script called at all. |
| 21:21.03 | pjensen00 | It's creating an AMQP queue to tell another service to create the Stasis app, and then when it's created it tells the AGI script to hand control over to Stasis. |
| 21:22.01 | pjensen00 | Is there a way to send a message to an external service before the default dialplan is executed? |
| 21:23.30 | pjensen00 | I'm not seeing a good way to prevent myself from having to do all the setup in the AGI script |
| 21:27.33 | DivideBy0 | I don't entirely follow you. Could you deliver the channel to a stasis app, do whatever you need, then send it back to the dialplan? |
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| 21:41.11 | pjensen00 | I'm wondering if there's a way I can create a stasis app upon a call hitting Asterisk |
| 21:41.33 | file | why do you need to create a stasis app unique to it? |
| 21:41.54 | pjensen00 | I use a different App for each call leg |
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| 21:44.52 | file | that sitll doesn't quite answer my question |
| 21:45.57 | pjensen00 | If one of my ARI process goes down, it's required to not affect any other processes also doing an ARI process |
| 21:46.35 | pjensen00 | aka, something happens that causes the web socket in one app to go down should have zero impact on other calls on the same server |
| 21:47.06 | pjensen00 | other than any other app bridged to the dead app sees the call terminating |
| 21:47.30 | file | nothing else springs to mind besides AGI then |
| 21:48.14 | pjensen00 | Wait, wait! Do you know if you can transfer one Stasis app to another Stasis app? |
| 21:49.18 | file | indirectly yes... you can continue to a spot in the dialplan |
| 21:50.16 | pjensen00 | Hrm. If that's the only solution, I think I can mad scientist this thing. |
| 21:50.28 | pjensen00 | Back to the thinking cube. |
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| 22:23.36 | mikealeonetti | What format does STREAM FILE need to be in? There doesn't seem to be much documentation about it, and I saw one script that uses GET FULL VARIABLE ${CHANNEL(audionativeformat)} to get the audio format to convert it into. What is really the best practice? |
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| 22:51.09 | [TK]D-Fender | mikealeonetti, best practice is to have your audio files int he same format as your call |
| 22:51.22 | [TK]D-Fender | mikealeonetti, and STREAM FILE can read any format * can. |
| 22:51.36 | [TK]D-Fender | "module show like FORMAT" <- |
| 22:51.58 | mikealeonetti | [TK]D-Fender: So should I be polling the channel and converting then? |
| 22:52.13 | [TK]D-Fender | No, you should simply convert all of them. |
| 22:52.21 | [TK]D-Fender | VS trying to do this "on demand |
| 22:52.47 | mikealeonetti | [TK]D-Fender: So the format will never change? Is it something that I set? |
| 22:53.38 | [TK]D-Fender | These are your calls... you are supposed to set the codec used for your calls |
| 22:54.40 | mikealeonetti | [TK]D-Fender: Oh that's kind of cool. What extension do I make the files then? 'Cause it says with STREAM FILE to exclude the extension. |
| 22:56.11 | [TK]D-Fender | You make ti to match the codec of your calls |
| 22:56.24 | [TK]D-Fender | * will pick the most appropriate format of that file automatically |
| 22:57.07 | mikealeonetti | Okay. That makes sense. |
| 22:57.14 | mikealeonetti | [TK]D-Fender: This sounds like fun. |
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| 23:28.31 | pchero | Hi.. Can I transfer channel to peer directly? without dialplan? |
| 23:37.04 | [TK]D-Fender | no |
| 23:37.52 | pchero | Clear. :) |
| 23:40.48 | [TK]D-Fender | I thought it was clear a week ago when you asked the same thing and got the same answer' |
| 23:43.09 | pchero | [TK]D-Fender: You're right. I just forgot that.. Now I'm writing about this to my blog. Because, I don't want to forget this again.. :P |
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