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| 06:13.37 | *** topic/#asterisk is Take the March 2019 Asterisk User Survey! https://goo.gl/forms/xL1VUHRsf95saly13 -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.28.0 (2019/7/25) 16.5.0 (2019/7/25), Security Only: 15.7.3 (2019/7/11); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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| 14:47.18 | HyP3r | Short question about the SIP-Protocol itself: I have here a Wireshark Trace of a Incoming Phone Call and the Caller *should* offer H.264-Video and now the question is: is it the phones fault to not accept the Video or is it the SIP-Server/Proxys fault? |
| 14:47.33 | HyP3r | I'll post the SIP INVITE and SIP OK |
| 14:50.31 | HyP3r | Here we are: INVITE: https://dpaste.de/eA3y OK: https://dpaste.de/znqh |
| 14:50.57 | HyP3r | I would say it is the phones fault because it does not acknowledge the H.264: "Media Description, name and address (m): video 0 RTP/AVP 99" |
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| 15:09.59 | Samot | Does the phone have h.264 codec enabled? |
| 15:10.10 | Samot | And in it's allowed list? |
| 15:11.43 | HyP3r | Samot: yes |
| 15:12.16 | Samot | I don't see it in the SDP. |
| 15:13.46 | HyP3r | Samot: https://drive.google.com/uc?id=1_AvR_zcnTIsZO_R0WIGyiMAn248QJg_i |
| 15:14.43 | Samot | So who is making the call? |
| 15:15.09 | HyP3r | The other side the INVITE is from the door intercom |
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| 15:16.40 | HyP3r | The screenshot is from the device with the number 2205996a |
| 15:17.27 | Samot | So the call connects and video never comes up? |
| 15:17.33 | HyP3r | Yeah |
| 15:17.49 | HyP3r | But in the INVITE there is a video offer |
| 15:18.02 | HyP3r | Media Attribute (a): rtpmap:99 H264/90000 |
| 15:21.37 | Samot | Yes but the Yealink is not offering it |
| 15:21.50 | HyP3r | Samot: so the Yealink device is not configured properly? |
| 15:22.07 | Samot | Well the screen shot shows it enabled. |
| 15:22.11 | HyP3r | Yeah |
| 15:22.14 | Samot | Does it offer video when it makes the call? |
| 15:22.25 | HyP3r | I didn't tested this case... |
| 15:23.01 | HyP3r | I read the Phones Administrator Guide but I guess I have everything confiured correctly |
| 15:23.13 | Samot | What model is this? |
| 15:23.24 | HyP3r | Yealink T58 |
| 15:23.27 | HyP3r | do you know Yealink? |
| 15:23.55 | Samot | Which T58? |
| 15:24.36 | HyP3r | I orderd this here: https://geizhals.de/yealink-sip-t58a-a1636876.html https://www.amazon.de/dp/B072LZWYPD |
| 15:24.42 | HyP3r | "Yealink SIP-T58A IP Phone " |
| 15:25.24 | Samot | And do you have the camera? |
| 15:25.30 | Samot | Or is this just for one way video? |
| 15:25.45 | HyP3r | No there is no camera on the Phone (not directly or over USB) |
| 15:25.49 | HyP3r | I want to have a one way video |
| 15:26.08 | HyP3r | Door Intercom (Mobotix T24) -> Yealink SIP-T58A IP Phone |
| 15:26.14 | Samot | OK then you can't test the other method. |
| 15:26.30 | Samot | Well you might be able to. |
| 15:26.34 | HyP3r | Do you think this is the problem? The Yealink does not have a camera so it doesn't accept video? |
| 15:26.43 | Samot | Make a call from the yealink to see if it offers video. |
| 15:27.02 | HyP3r | I'll do that at monday I already drove home from the customer... |
| 15:37.11 | *** join/#asterisk newbeeasterisk (c3b5d3c6@195.181.211.198) |
| 15:39.01 | newbeeasterisk | Hi for all!!!! I have issue with asterisk gui....I can't login via web interface while continuously LOOP with backup. Any help please? |
| 15:41.21 | Samot | Asterisk doesn't have a GUI. |
| 15:43.41 | newbeeasterisk | ....then what is this please? ---> http://svn.digium.com/svn/asterisk-gui/branches/2.0/ gui |
| 15:45.18 | newbeeasterisk | Unfortunately I see always this screen: https://ibb.co/t4H3cXV |
| 15:47.18 | Samot | That was an old project that never really got anywhere. |
| 15:47.46 | Samot | It's also like 10 years old. |
| 15:47.53 | Samot | So it wouldn't apply to anything current. |
| 15:48.27 | newbeeasterisk | Thank you, then what's now please? (I'm totally new, now I try to learn, please help) |
| 15:54.10 | Samot | FreePBX is the main choice generally when looking for GUI control over Asterisk. |
| 15:56.29 | newbeeasterisk | Thank you. FreePBX is up-to-date please? |
| 15:56.49 | Samot | Yes. |
| 15:59.36 | newbeeasterisk | Anyway, my goal to setup a phone survey system (like this: https://voicent.com/phone-survey/), I have usb 4G modems, I want to learn, but I'm ready to pay for a ready-made solution also. Anyone interested please? |
| 16:04.30 | TandyUK | sorry hell no, and if a system like that ever phones me up, it will be sworn at prfusely until it gives me a fking human |
| 16:21.41 | newbeeasterisk | YES! You are right, our project going on this (right) way, systems call play-back a very short (10-20 sec.) marketing message (it will be call ONLY SELECTED numbers) and will be ask: press "1" if you want human, or press "2" if this offer not interested for you. In this case it's an absolutely "friendly" application. |
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| 19:15.01 | avb | guys, who knows what is the settings similar to pedantic=no would be in pjsip? |
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| 19:32.45 | Samot | Not really sure there is one. |
| 19:33.15 | avb | yeh, seeems its not implemented :-/ |
| 19:33.30 | file | PJSIP doesn't have such functionality |
| 19:33.56 | avb | got a problem that genesys is sending me BYE with a wrong branch |
| 19:34.16 | avb | any other ways to disable branch check on bye in pjsip? |
| 19:34.22 | file | no. |
| 19:34.28 | avb | huh :) |
| 19:34.52 | avb | seems need to move things to chan_sip :( |
| 19:35.03 | Samot | This doesn't seem like an Asterisk issue. |
| 19:35.46 | file | the BYE would be a new transaction though and thus the branch should be different ... |
| 19:35.47 | avb | it does not, but other guys dont want/cant fix it :) |
| 19:36.12 | Samot | Well there is what file just said too. |
| 19:36.23 | Samot | New transactions (request) have their own branch. |
| 19:36.36 | avb | so asterisk returns 481 and keeps the second leg up forever |
| 19:36.55 | Samot | Are the from/to tags correct? The Call-ID? |
| 19:37.14 | Samot | Those are three things used for a dialog |
| 19:37.26 | avb | maybe i have overlooked and its not a branch, sorry |
| 19:37.50 | avb | i havent saved a trace ... :) |
| 19:38.50 | avb | thanks guys |
| 19:39.04 | avb | you haved me hours of googling :) |
| 19:39.06 | avb | saved* |
| 19:44.44 | avb | Samot: you are correct, Invite has To: <sip:xxx@yyy:5060> and bye has To: <sip:xxx@yyy:5060>;tag=as1fcae397 |
| 19:44.59 | avb | i found a trace |
| 19:45.14 | Samot | The initial INVITE will always lack a to tag. |
| 19:45.33 | avb | huh |
| 19:45.38 | avb | rest seems is the same |
| 19:45.59 | Samot | When send a call from my phone to the PBX, the to tag doesn't exist. |
| 19:46.34 | Samot | It's updated in the transaction. |
| 19:47.52 | avb | callid is in the uuid format |
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| 21:39.27 | forgotmynick | i have asterisk installed on a remote dedicated server. in my home i have a landline. is there a device i can use to connect the landline to the remote pbx server? |
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