| 04:32.40 | *** join/#asterisk infobot (ibot@c-174-52-60-165.hsd1.ut.comcast.net) |
| 04:32.40 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.32.0 (2020/03/12) 16.9.0 (2020/03/12) Standard: 17.3.0 (2020/03/12); 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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| 08:18.15 | yang | Hello, I have a problem with incoming calls. I am trying to call +38617774515 which is a number allocated/forwarded to me by the SIP provider (localphone.com). The phone on my side doesn't ring. It can also be a firewall issue? I can mention that I am able to do outgoing calls through the same SIP account (provider). Here is the debug log http://paste.debian.net/hidden/345b13fe/ |
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| 08:36.01 | yang | From: "38669944499" <sip:38669944499@95.211.119.240>;tag=11-2D00F42F-5EA29DAF00012FE0-D1D9D700 |
| 08:36.04 | yang | To: <sip:38617774515@localphone.com> |
| 08:36.19 | yang | I can see here that call is being registered, but not processed successfully |
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| 08:54.45 | chris^ | does anyone know any asterisk and/or restcomm contractors? Need some install and config help |
| 08:59.12 | yang | if you type "asterisk consultants" in (google) search, you will find many websites listing these companies |
| 08:59.38 | chris^ | I have, none that do both |
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| 13:59.07 | *** mode/#asterisk [+o kharwell] by ChanServ |
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| 14:12.34 | igcewieling | It sure would be nice if Asterisk got RTCP right. I have a call with "1330" percent packet loss. |
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| 14:34.20 | allizom | Hi all, I'm trying to have video conferences with asterisk, but when I use video_mode=sfu I don't actually get any video. This is my configuration: https://paste.centos.org/view/raw/4e19c93b and log: https://paste.centos.org/view/raw/1499c868 When I use videoMode=follow_talker I correctly get video from the current speaker. |
| 14:35.12 | file | SFU requires the client to allow multiple simultaneous video streams |
| 14:35.19 | file | yours rejected it |
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| 14:35.48 | allizom | file: I'm using a client which can receive multiple videos |
| 14:35.55 | file | it still rejected it |
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| 14:36.26 | file | this is just re-negotiating to add an additional video stream: |
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| 14:36.29 | file | https://www.irccloud.com/pastebin/CZO1yzQ8/ |
| 14:36.47 | file | Linphone kept the already negotiated stream, but rejected the newly added one: |
| 14:36.50 | file | https://www.irccloud.com/pastebin/QaU55M7B/ |
| 14:37.16 | allizom | hm, that's strange, I've had video conferences with linphone |
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| 14:37.31 | file | they may have been a mixed single video stream |
| 14:37.47 | allizom | it was not |
| 14:37.52 | allizom | they wew separate stremas |
| 14:37.56 | file | then I do not know |
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| 14:40.34 | allizom | file: is this the line where linphone rejects video? a=inactive |
| 14:40.41 | file | yes |
| 14:40.47 | allizom | I see |
| 14:41.09 | allizom | let me find again the session from a video conference I've had |
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| 16:18.41 | Slade | Reinhilde, not expensive, people either do it or not |
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| 16:26.01 | Samot | The caveat: Not doing it can result in it being _really expensive_ |
| 16:33.07 | igcewieling | Any comments about Sangoma Meet? |
| 16:33.45 | Samot | It looks pretty decent. |
| 16:34.10 | igcewieling | The screen sharing is most interesting to me. |
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| 16:36.25 | igcewieling | I think I'll wait until the rush is over. I keep getting page is not responsive messages. |
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| 16:59.23 | MLC | For those that have been following my saga, I finally have both Bria and Digium phones connecting remotely with both TLS signaling and media encryption. Life is good! |
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| 17:27.34 | yang | Hello, I have a problem with incoming calls. I am trying to call +38617774515 which is a number allocated/forwarded to me by the SIP provider (localphone.com). The phone on my side doesn't ring. It can also be a firewall issue? I can mention that I am able to do outgoing calls through the same SIP account (provider). Here is the debug log http://paste.debian.net/hidden/345b13fe |
| 17:27.40 | yang | I can also see the call being received in "debug" From: "38669944499" <sip:38669944499@95.211.119.240>;tag=11-2D00F42F-5EA29DAF00012FE0-D1D9D700 - To: <sip:38617774515@localphone.com> |
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| 18:04.18 | Slade | Samot, haha |
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| 18:09.09 | qakhan | hi, anyont tried google speech to text with * ? |
| 18:09.18 | qakhan | anyone* |
| 18:18.04 | MLC | I've used it with asterisk recordings, but not directly integrated with Asterisk |
| 18:22.22 | qakhan | i have integrated it with asterisk, |
| 18:22.33 | qakhan | i record the voice and send it to google. |
| 18:23.20 | qakhan | but i am looking for live call streaming and get the text from speech |
| 18:30.42 | Slade | i do not approve of the * abbreviation for asterisk software. |
| 18:31.23 | joepublic | even worse is what >50% of "asterisk training videos" call it, "ass tricks" |
| 18:32.55 | Slade | I'd guess that "ass tricks" is more common in every day speech |
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| 18:45.46 | qakhan | ok lets come back to topic |
| 18:46.09 | qakhan | anyone who worked on google speech to text with asterisk |
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| 18:59.10 | Samot | I'm not sure how you would "live stream" this over to google. |
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| 19:03.19 | qakhan | e.g when we call comcast. their IVR recognize voice on run time. |
| 19:03.56 | Samot | That's irrelevant. |
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| 19:04.14 | qakhan | this the example i am giving you |
| 19:04.21 | qakhan | i am trying to do same thing |
| 19:04.27 | Samot | Does Comcast use Google? |
| 19:04.33 | Samot | Do they have something internal? |
| 19:04.34 | qakhan | I dont know |
| 19:04.37 | Samot | Exactly. |
| 19:05.00 | qakhan | comcast IVR is an example, i am refering to |
| 19:05.03 | Samot | My point was, how would you get the live stream from ASterisk to google? |
| 19:05.37 | qakhan | incoming call on asteriks is live audio stream |
| 19:06.39 | qakhan | comcast IVR ask, "say you account # or serivce address" |
| 19:07.16 | Samot | OK so you would have to record that, send it to google and get a response |
| 19:07.36 | qakhan | that is what i am doing |
| 19:07.41 | Samot | OK |
| 19:07.56 | qakhan | but its lil slow |
| 19:08.02 | Samot | So again, to your comment about "live streaming to Google" I said I'm not sure how you would do that. |
| 19:08.22 | qakhan | understand |
| 19:08.44 | Samot | What you are doing now is probably the best method. |
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| 19:09.14 | qakhan | i get the call and send to agi->record |
| 19:09.43 | qakhan | i wanted to know if someone else is doing same thing or not |
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| 21:13.14 | velix | ROFL, this trunk has some fun: "SIP/2.0 100 trying -- your call is important to us" |
| 21:13.31 | MLC | LOL |
| 21:15.30 | velix | I only need "type = identify" for incoming calls, right? |
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| 22:05.11 | Maria | Hi. I have 2 problem with asterisk last rev. 1) Software on mobile work properly. Softphone in laptop say Idle. I can call from laptop but I can't receive any call. |
| 22:05.25 | Maria | wheri is the problem? |
| 22:05.35 | *** part/#asterisk MLC (~MLC@63.249.40.11) |
| 22:07.53 | jsmith | Maria: Is the softphone properly registered to the PBX? If the softphone isn't registering properly, the PBX doesn't know where to send the calls to. |
| 22:08.41 | Maria | But I can call with the softphone and the registration it is the same of the mobile |
| 22:09.58 | Samot | Is the mobile on the same network as the laptop? |
| 22:10.28 | Maria | yes |
| 22:10.56 | Samot | You are using chan_pjsip? |
| 22:12.13 | Maria | Yes, in pjsip.conf there are only few row. |
| 22:12.13 | Maria | [101](banana-voip) |
| 22:12.13 | Maria | secret=pass101 |
| 22:12.13 | Maria | callerid="Fabrizio_Cell" <101> |
| 22:12.24 | igcewieling | calls from clients to server have NOTHING to do with registration. |
| 22:12.53 | Samot | And you have thel max_contacts set to 2? |
| 22:13.05 | Samot | In the aor section? |
| 22:13.42 | Maria | the problem is that I can call from the laptop to mobile, but not from mobile to laptop. I'm going to check |
| 22:14.00 | Samot | That wasnt my question |
| 22:14.26 | Maria | Samot: aor section in pjsip.conf? |
| 22:14.33 | Samot | Yes. |
| 22:14.45 | Maria | I don't have a aor section |
| 22:15.22 | Samot | Thats a problem |
| 22:18.14 | Samot | It's a rather important thing to have. |
| 22:18.44 | Maria | Samot: Ok. I need to search how to wrtite the correct aor. There is a stupid example? |
| 22:19.13 | Samot | The Wiki, the sample config |
| 22:19.41 | Maria | Samot: ENDPOINT CONFIGURED AS A TRUNK, INBOUND AUTH AND REGISTRATION parts? |
| 22:20.23 | Samot | Same concept. |
| 22:21.15 | Maria | Samot: Ok, thanks. I'll try to undertand that example. |
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