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| 05:57.45 | cepxuo | Calls cancelled by caller during the dialing phase, are shown in Linphone as simply past calls, not missed ones. |
| 05:57.45 | cepxuo | Is this my dialplan / setup or an Asterisk issue? How can I get Asterisk to send cause=487? |
| 05:58.04 | cepxuo | I thought this is an Linphone issue, but Sylvain says it's on my PBX side: |
| 05:58.04 | cepxuo | https://github.com/BelledonneCommunications/linphone-android/issues/832#issuecomment-557474864 |
| 05:58.04 | cepxuo | > The CANCEL message has a Reason header with Q.850 protocol and cause 0, which doesn't mean call has been missed (should be 487). |
| 05:58.35 | cepxuo | already tried to ask in asterisk-users mail list without success |
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| 06:48.45 | malarinv17 | hi |
| 06:49.14 | malarinv | cool |
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| 07:57.37 | sibiria | gry: right, but give 'config list' a try, as i mentioned |
| 07:58.28 | sibiria | so if you just add users.conf, it (and the modules that use it) should show up in that list |
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| 08:00.16 | Ravenheart | hey guys |
| 08:00.58 | Ravenheart | i have all my local (in the office) extensions under a context called "internal" |
| 08:01.12 | Ravenheart | when i'm dialing out is there a way to change the context? |
| 08:02.20 | Ravenheart | https://hatebin.com/obwqrjqnzq |
| 08:02.24 | Ravenheart | my dialplan |
| 08:03.10 | Ravenheart | bascially i need a way (with AMI) to detect a new call to external phones and internal ones |
| 08:03.24 | Ravenheart | so i can filter out all the internal to internal calls |
| 08:03.56 | Ravenheart | sadly CHANNEL(context) is read only |
| 08:12.25 | Ravenheart | i have the weird suspicion the channel is predominately american and everybody is asleep :) |
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| 08:38.40 | Ravenheart | and not even Set(CHANNEL(accountcode)=bla) works :( |
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| 10:20.32 | GeneralSpongebob | Hello. What is the relationship between Peer and Device in Asterisk? I see there are two separate events, PeerStatus and DeviceStateChange, but I can't see anything in the documentation to define what a Peer or a Device is to Asterisk. |
| 10:22.56 | file | PeerStatus refers to the channel driver status, DeviceStateChange refers to the core view and provides more information |
| 10:25.40 | GeneralSpongebob | I see. Thank you for explaining |
| 10:26.14 | file | for example a peer can be registered, but from a core device perspective they may be in use because they're on a call |
| 10:28.06 | GeneralSpongebob | Thanks, it's a little confusing but now I've got a better idea of the difference I can go over my AMI logs again :) |
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| 12:22.54 | zicada | If one wanted to create a webapp that can both send and recieve phones to Asterisk, how would you recommend one went about that ? |
| 12:23.01 | zicada | webRTC ? |
| 12:23.28 | zicada | /s/phones/phonecalls |
| 12:23.57 | file | that is generally what most people do these days. |
| 12:26.47 | zicada | and then authentication is done with something like coturn, is that right ? |
| 12:26.57 | zicada | in order to do rates etc |
| 12:29.20 | zicada | Would i still be able to use ARI to combine legs from my regular SIP upstream and a webRTC client into a conference ? |
| 12:30.30 | file | authentication of what? SIP is what is implemented in Asterisk for signaling using WebRTC, SIP is SIP |
| 12:30.43 | file | and Asterisk doesn't treat WebRTC calls any different than normal calls |
| 12:30.53 | zicada | all right, good to know |
| 12:31.18 | zicada | Like, in this webapp, you would be able to buy a real phone number and immediately use it |
| 12:31.55 | zicada | Like, how will i know the client making the request are the ones paying for the service |
| 12:32.14 | zicada | i saw somewhere they use these STUN/TURN servers for this ? |
| 12:32.33 | file | nope |
| 12:32.42 | file | STUN is used for discovering your external IP address and port and for ICE |
| 12:32.51 | file | TURN is for media relay in case behind NAT and direct is not possible |
| 12:35.39 | zicada | right i see. And then WebRTC as a standard makes use of these services then, to get around problems with NAT |
| 12:36.59 | file | yes. |
| 12:37.00 | zicada | But would you have Asterisk itself do authentication over SIP then ? |
| 12:37.25 | file | SIP is SIP, and how it's all deployed and such is up to you |
| 12:37.39 | zicada | yeah i mean, if you were gonna do something like this, how'd you consider handling that :) |
| 12:38.02 | file | haven't given it any thought. |
| 12:38.04 | zicada | its an open port to the internet, not a trunk from an upstream which im used to, new territory :) |
| 12:43.17 | Ravenheart | i don't need to specify an acl.conf in manager.conf users if i've specified deny/permit entries for each user right |
| 12:43.22 | Ravenheart | asterisk 16 |
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| 16:02.52 | Reinhilde | gry: :O |
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| 18:16.27 | dangmoo | I have 2 different VPS servers with Asterisk installed. |
| 18:16.27 | dangmoo | When I try to call from one of them to a SIP URI on the other (using a softphone) I get this error: |
| 18:16.27 | dangmoo | Failed to authenticate on INVITE to '"username"... |
| 18:16.28 | dangmoo | but when I set the CALLERID(num) variable just before the Dial app everything works fine. |
| 18:16.29 | dangmoo | In both of the servers 'allowguest' is enabled by default. |
| 18:16.30 | dangmoo | My question is which setting is responsible to this behavior? Why Asterisk rejects anonymous calls by default ? How would I enable/disable anonymous calls if I wanted? |
| 18:16.31 | dangmoo | Where can I read more about this? |
| 18:25.21 | zicada | CALLERID is required either way i thought |
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| 18:26.24 | zicada | if you want to write CDR, you want to include CALLID |
| 18:26.58 | zicada | would be nice if there was a standard |
| 18:28.16 | zicada | 3-letter namas will require access |
| 18:28.57 | zicada | the security-community werent really involved im told |
| 18:53.56 | Ravenheart | which codec gives the best clarity and volume |
| 18:54.10 | Reinhilde | ? |
| 18:54.28 | Reinhilde | Ravenheart: clarity, anything uncompressed, hotly chased by opus. volume, not influenced by codec. |
| 19:03.41 | Ravenheart | which one do you use in your asterisk installations |
| 19:05.41 | Reinhilde | i tend to use 711 for narrowband and 722 for wideband |
| 19:12.04 | igcewieling | same here, ulaw for narrowband, g722 for wideband. no need for other codecs. |
| 19:13.23 | Reinhilde | I've probably unintentionally used gsm before. That's fine, if you don't want music. |
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| 19:43.13 | zicada | it depends on alot of things |
| 19:43.36 | zicada | if you mostly do calls in one country and their tech allows a really high quality, obv you use that |
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