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| 02:40.14 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.23.1 (2018/09/20) 16.0.0-rc3 (2018/09/20), Standard: 15.6.1 (2018/09/20); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 15:48.44 | _AxS_ | hey all -- i'm making the switch to voip at my residence, and have been messing about with asterisk for a little while now. I've got a few different phones (sip, iax-softphones), but I want to make them all behave as if they're on the same POTS line -- not just all ring when a call comes in, but also conference into an existing call when picking up another handset.. is the latter feasible? |
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| 15:50.13 | igcewieling | you can do conferencing, just don't expect to put a call on hold at one handset and pick it up from another handset without doing anything else. |
| 15:50.31 | _AxS_ | its not so much that, rather I |
| 15:51.39 | _AxS_ | ... i'm hoping that if another person wants to talk on the call as well it can be as simple as them picking up the handset, or -maybe- dialing an extension to join. but the key is to be able to do this on any call |
| 15:52.40 | igcewieling | It doesn't work like old multi-button PBXs. |
| 15:53.57 | _AxS_ | I guess there's no way to make it work like that either then... I was thinking perhaps if it were possible to always put an incoming or outgoing call into a conference, then others could join in as-needed with very little in the way of special treatment. |
| 15:54.17 | MrTAP | _AxS_: look at SCA Shared Call Apperance |
| 15:54.27 | _AxS_ | MrTAP: thx! will google now |
| 15:54.48 | igcewieling | https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=4816923 |
| 15:54.56 | igcewieling | SCA ^^^ |
| 15:55.22 | MrTAP | SLA yeah that's what I was thinking of |
| 15:56.20 | igcewieling | I think you'll find it would take a lot of work to emulate key systems with Asterisk and I doubt anyone else here has set this up like in the wiki |
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| 16:03.52 | _AxS_ | ugh. maybe i should just get an ATA and use a regular wireless handset on it, handle all this common-line stuff downstream. |
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| 16:04.18 | _AxS_ | s/use a /use a set of / |
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| 18:26.46 | Kobaz | what's a good way to debug incomplete MixMonitor recordings |
| 18:28.10 | Kobaz | i have a 12 second .sln |
| 18:28.22 | Kobaz | when the talk time on the call is 1:49 |
| 18:28.34 | Kobaz | ...no errors in the logs |
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| 19:25.43 | n8nate | Samot: Thanks for the recommendation to get myself moving to 13.23.1 in our conversation yesterday. I got that onto a test server and began working on my authentication issues (I was n8n8 yesterday...). In some of my research one article I found indicated that 401 is the normal challenge response to SUBSCRIBE, and 407 is normal for INVITE. My PBX (both 13.172 and 13.23.1) is sending my endpoints 401 to INVITE's... Is that a misconfig on my side, or, just how |
| 19:25.43 | n8nate | PJSIP does it? |
| 19:26.06 | n8nate | that's 13.17, not 13.172 |
| 19:26.43 | Samot | SUBSCRIBE/REGISTER/INVITE all involve an auth process. |
| 19:26.54 | Samot | The process is the same for each of them. |
| 19:27.32 | Samot | The initial request is sent *without* any auth details in it. No user, domain, password. Nothing. |
| 19:28.18 | Samot | If you are not challenging the fact none of this exists and/or are allowing auth via the source IP and validate against that..there is no 401/407 sent back. |
| 19:28.32 | n8nate | understood, I'm wondering if there is an implementation issue where they'll auth to a 407 after invite, but not a 401. |
| 19:28.51 | n8nate | when I turn on auth, as you saw yesterday, the endpoints just ack'ed |
| 19:29.04 | Samot | If you are challenging it, you send back a 401 or 407. Either of those replies should tell the requesting endpoint to re-send a new request *with* auth headers. |
| 19:29.52 | n8nate | My quick skim through the RFC suggested that proxy's challeng invite with 407. |
| 19:29.58 | n8nate | my pbx is sending 401 |
| 19:30.00 | Samot | Yes, but your SUBSCRIBE's also showed you challenging them correctly, the user responding correcting and instead of the proper 200 OK response you're sending a 500 Server Error response. |
| 19:30.13 | Samot | Asterisk is not a proxy. |
| 19:30.17 | n8nate | understood |
| 19:30.30 | Samot | It is a B2BUA. It uses 401 Unauthorized challenges. |
| 19:30.56 | n8nate | I didn't see in the RFC the auth process for the B2BUA portion, I may not have gotten far enough along. |
| 19:31.10 | Samot | It's the same process. |
| 19:31.21 | Samot | 401 OR 407 are acceptable replies. |
| 19:32.53 | n8nate | Yes, I'm just wondering if it is worth the trouble to work with the MFG on why they ACK the 401, or if I should be looking deeper on my side as to why I'm not sending 407... However, now that you point out the 500 response to the 200 from the UA... that might be the source of wy * sends 401 instead of 407 |
| 19:33.11 | Samot | GD. |
| 19:33.13 | Samot | Stop. |
| 19:33.19 | Samot | There is not issue with it being a 401. |
| 19:33.31 | Samot | The phone would have to be able to accept EITHER of those responses. |
| 19:33.43 | n8nate | Then I should work with Grandstream |
| 19:33.44 | Samot | Asterisk sends a 401 |
| 19:33.50 | Samot | Not 407. |
| 19:33.55 | Samot | Because it's not a proxy |
| 19:34.11 | Samot | So 401 is the proper reply to those requests. |
| 19:34.47 | Samot | When a 407 is returned because it is a proxy it means there is a hop between the device and where it really wants to go... |
| 19:35.07 | Samot | And that hop is saying, "You have to auth with me first before getting through |
| 19:35.09 | Samot | And that hop is saying, "You have to auth with me first before getting through" |
| 19:35.13 | n8nate | As I said, I'm trying to figure out where to spend my time. It follows that there is an implementation issue in the Grandstream FW. |
| 19:35.21 | Samot | Nope. |
| 19:35.38 | Samot | People have 100's of Grandstreams deployed with Asterisk/FreePBX |
| 19:35.49 | Samot | Many running Asterisk 13 or higher at this point. |
| 19:35.52 | Samot | And using PJSIP. |
| 19:35.52 | n8nate | If they respond to the 401 with ACK and do not try to auth, where is the problem ten? |
| 19:35.59 | n8nate | then* |
| 19:36.04 | Samot | Based on what I've seen so far... |
| 19:36.14 | Samot | This could be more related to your setup than the phones. |
| 19:36.33 | Samot | Your setup is clearly showing issues with how it is configured. |
| 19:36.35 | n8nate | I have no doubt my config is messed up. I'm trying to figure out where to look |
| 19:36.49 | Samot | Well first... |
| 19:37.00 | Samot | Dump Real-Time. It is pointless. |
| 19:37.15 | Samot | Test all this without Real-Time. |
| 19:37.20 | Samot | Use the static configs. |
| 19:42.10 | n8nate | So the idea is to replicate, or not, with a trivial static config. (full environment from realtime to static is crazy-non-trivial). Then, If I am able to duplicate the problem, then my testing should be pretty clear cut from that point. If I cannot duplicate the problem, we're using this as the case for me migrate off realtime, or, debug the differences in how * handles config from realtime vs static config? |
| 19:47.28 | _AxS_ | hey all, have another question -- i get these messages every so often on my sip trunk: "handle_response_register: Got 423 Interval too brief for service XXXX, minimum is 3600 seconds". I've set min/max/default expiry to 3600, but in 'sip show registry' it lists the refresh at 3585 which i think is what's triggering this as it's less than 3600 ... |
| 19:47.37 | _AxS_ | ..or am i looking at entirely the wrong place? |
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| 20:07.22 | darkunderlord | Hey everyone. So I've got what seems like a weird situation. I am taking some paging processes from an 11 SIP server to a 13 PJSIP server. I'll post the code in a minute on pastbin, but maybe this will ring a bell.... Normal calls are fine and only ring once. The page (basically to only one device to play audio for visual voicemail app via callfile) actually makes two duplicate calls to the phone. I've gone through my |
| 20:07.23 | darkunderlord | 3:57 PM dialplan over and over and am not new to Asterisk by far. Just driving myself nuts. |
| 20:07.23 | darkunderlord | 3:57 PM Anyone get duplicate calls like this before? |
| 20:07.23 | darkunderlord | 4:06 PM Here's the log: https://pastebin.com/1CVvjLjZ , relevant dialplan: https://pastebin.com/ajJUEWQ7 , console output, even though log has it too: https://pastebin.com/RdftRz78 |
| 20:09.40 | darkunderlord | also call file: https://pastebin.com/6sYnyuMB |
| 20:13.15 | [TK]D-Fender | turn off the core debug. |
| 20:13.21 | [TK]D-Fender | It's a psychotic amount of noise |
| 20:15.16 | darkunderlord | i can post the regular log if you'd like lol |
| 20:22.46 | darkunderlord | [TK]D-Fender: so..any help or sound familiar? |
| 20:23.31 | [TK]D-Fender | I haven't seenc lean debug yet |
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| 20:30.17 | darkunderlord | [TK]D-Fender: so you want the default messages file output? What do you mean by lean. |
| 20:30.50 | darkunderlord | LOL is it something I said? ;) |
| 20:49.50 | _AxS_ | darkunderlord: it's async in here. just pastebin the log with core debug off, i'm sure they'll get to it at some point (or someone else will) |
| 20:56.40 | darkunderlord | Debug with core off. :) https://www.irccloud.com/pastebin/g5JpRUeO/ |
| 20:56.48 | file | darkunderlord: the call file is doing exactly what you told it to - dial extension 9999999 in the default context using a Local channel, and once answered send it to extension 9999999 in the default context |
| 20:57.08 | file | so effectively you are telling both sides of the Local channel to go to the same place and do the same thing |
| 20:58.04 | darkunderlord | but extension 312 is the one I want to ring :) Hmm, I need to make sure I put the header on to auto answer and play the file through the phone. I have it working on 11 with SIP. Not exactly like this, but very similar with same 99999 extension, etc |
| 20:58.17 | darkunderlord | Thanks for the reponse @file |
| 20:59.05 | darkunderlord | in teh SIP version, I'm able to add the SIP Header auto answer to the callfile, and use the Playback app |
| 20:59.58 | darkunderlord | Can I somehow do the sip header I want in the callfile? I tried with the PJSIP style and it didn't work. It kept saying that a PJSIP channel wasn't available since I had to playback on the Local channel |
| 21:00.20 | file | no, you can't do it from the call file |
| 21:00.26 | file | you have to send it to a Local channel |
| 21:00.33 | file | but you can't send it to the same place in both cases |
| 21:01.11 | darkunderlord | how can I send it to one PJSIP extension? I'm just playing the voicemail file through the phone |
| 21:01.34 | file | you send it to a dialplan extension using a Local channel that dials the PJSIP extension |
| 21:02.12 | darkunderlord | did you see my commented dial command? I tried that too instead of the Page and it still did two calls. What am I doing wrong? |
| 21:02.35 | file | a call file does this: |
| 21:02.39 | file | it dials a channel |
| 21:02.51 | file | once answered the channel is sent to a place in the dialplan or an application |
| 21:03.03 | file | you have told it to send it to the SAME place that you originally dialed |
| 21:03.05 | file | you need to send it elsewhere once answered |
| 21:03.24 | darkunderlord | oh. So channel and extension are wrong LOL |
| 21:03.30 | darkunderlord | I thought it seemed redundant |
| 21:03.32 | darkunderlord | ;) |
| 21:03.33 | file | yes |
| 21:03.39 | darkunderlord | I'll try that. Thanks! |
| 21:15.21 | darkunderlord | @file Thanks so much for the help! Got it working. Have a great weekend... |
| 21:15.33 | file | cheers |
| 21:20.54 | Samot | n8nate: I just saw your reply. The answer is, yes. |
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