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| 00:16.01 | *** 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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| 01:20.22 | dongs | sip |
| 01:20.28 | dongs | err, i mean. sup |
| 01:21.13 | dongs | i've got quite a problem, trying to redirect a call incoming from one voip provider to my sellfone (which is also accessible via another voip provider) but asteris tries to do "local bridging" and there's no audio |
| 01:21.21 | dongs | all respective peers/etc have directmedia=no and such |
| 01:22.09 | devdvd | Hi all. I have an extension that toggles a custom device state. It works but I was just wondering if there's a better/cleaner way. Here's the code https://gist.github.com/devblueray/e223935980268dd92dc8cdffc5563f9b |
| 01:22.47 | dongs | devdvd: what do you expet to clean up in it? dialplan scripting isnt exactly pretty |
| 01:22.53 | dongs | looks allright |
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| 01:23.08 | devdvd | nothing really, I was just wondering if there was a better way |
| 01:33.29 | dongs | lokos ok to me |
| 01:33.40 | dongs | oh well, i guess my redirect thing i just solve by registering sellfone directly |
| 01:33.49 | dongs | instead of dialing it through another provider |
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| 03:19.00 | dongs | what do people use for sip server these days if not asterisk? |
| 03:19.11 | dongs | anything else around these days? im still running same asterisk from like 10+ years ago |
| 03:19.58 | dongs | quick google search suggests no |
| 03:34.21 | Samot | Asterisk is not a sip server. It's a B2BUA. |
| 03:51.49 | dongs | not what i asked but oaky |
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| 03:55.01 | Samot | well they are two different things. |
| 03:55.08 | Samot | I run SIP servers and Asterisk. |
| 03:55.21 | Samot | Because Asterisk can do things the SIP server can't. |
| 03:55.25 | Samot | And vice versa. |
| 03:56.46 | Samot | However, in comparison there is FreeSWITCH and even 3CX has a free version these days. |
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| 12:14.17 | DanielYK | My asterisk is behind a NAT and I have problems with the RTP stream. I use pjsip. I hear silence if I do no port-forwarding on my router for the RTP port-range of my asterisk (10000-10200). Which settings are important to work without port forwarding? |
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| 12:39.24 | sibiria | RTP is stateless when it comes to each end receiving audio |
| 12:39.34 | sibiria | you *need* to forward ports if you want to receive audio from the other end |
| 12:42.35 | sibiria | ie. the two nodes (your asterisk setup, and the remote) will tell eachother "send your audio to me towards this port" |
| 12:42.57 | sibiria | and if you're not directing that traffic towards asterisk then there will be certain silence |
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| 13:43.45 | file | it's best to, at least |
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| 13:57.28 | DanielYK | It is a bit confusing, if the endpoint is not behind a second NAT, it works without any problem. Also, isn't it any security risk to forward the complete UDP Port range? Because: 1) its an big range (even if I shrink it to maybe 100 ports) 2) Incoming UDP packages are forwarded in case of no SIP call exists. Maybe the ports are used for something else. |
| 13:58.40 | igcewieling | DanielYK: Generally the outgoing audio will trigger the correct NAT translations which will cause incoming audio to work without port forwarding. |
| 13:59.21 | file | "it depends" |
| 13:59.27 | igcewieling | Indeed. |
| 13:59.38 | sibiria | outgoing from port X to port Y will not set a state for port X +/- 1 |
| 13:59.47 | DanielYK | In normal cases, yes. But if the endpoint itself is behind a second NAT, the outgoing audio will not reach asterisk and asterisk cannot forward it to my ISP |
| 13:59.59 | igcewieling | what makes you think the other direction is port + 1? |
| 14:01.08 | DanielYK | Is there another solution for this case? E.g. Asterisk sends dummy packages to open a NAT session? |
| 14:02.04 | sibiria | igcewieling: if he had been using rtp_symmetric he'd have an open state and would not have a one-way audio problem |
| 14:02.47 | DanielYK | I use rtp_symmetric. |
| 14:03.00 | igcewieling | that I can agree with. |
| 14:03.14 | sibiria | are you using nat and not bi-nat? |
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| 14:03.53 | Samot | What is the router? |
| 14:04.01 | DanielYK | ISP <-> NAT <-> Asterisk <-> NAT <-> Endpoint: Does not work |
| 14:04.01 | sibiria | or, is your firewall explicitly configured to not keep state for outgoing UDP traffic? |
| 14:04.07 | DanielYK | ISP <-> NAT <-> Asterisk <-> Endpoint: Works |
| 14:04.27 | Samot | Wait... |
| 14:04.35 | Samot | The endpoint that is having an issue is remote to the PBX? |
| 14:04.41 | DanielYK | Yes |
| 14:04.50 | Samot | OK and what do they have where this endpoint is? |
| 14:04.53 | Samot | What router? |
| 14:05.28 | file | Playback(silence/1) or rtp_keepalive option may do something |
| 14:05.42 | DanielYK | I don't know if you know that kind of router, Fritz!Box |
| 14:05.52 | Samot | Oh dear god. |
| 14:06.02 | Samot | Do they have one way audio issues on outbound calls? |
| 14:06.08 | file | your scenario isn't double NAT, it's both sides being behind NAT which is different |
| 14:06.12 | Samot | Or just on inbound calls? |
| 14:06.28 | Samot | Plus Fritz!Box's are horrid. |
| 14:06.31 | file | (well - Asterisk behind a NAT with no ports forwarding, and a remote endpoint behind NAT) |
| 14:06.32 | DanielYK | Only on outbound calls |
| 14:06.51 | Samot | So when the endpoint makes a call, it has no incoming audio? |
| 14:07.53 | DanielYK | Currently in one special case: If I call a number with an IVR |
| 14:08.06 | DanielYK | I can call other endpoints and other numbers without problems |
| 14:08.40 | Samot | And no other endpoints have an issue when calling an IVR? |
| 14:08.55 | sibiria | file: i've been asking myself if that wouldn't be a useful feature in asterisk (playing some silence just to trigger RTP traffic to get things going), because i keep running into trunk providers where we lose about 1 second of initial audio on all calls |
| 14:09.09 | sibiria | so i have to prod them with 0.1 seconds of very low, almost silent white noise, as the first thing in the dial plan |
| 14:09.12 | file | sibiria: silence incurs a penalty |
| 14:09.15 | DanielYK | If I call exact this IVR with and endpoint in the localnet, then it works propaply |
| 14:09.20 | file | it requires a translation path |
| 14:09.41 | file | so it's just easiest to do a Playback of silence if you want to |
| 14:09.57 | Samot | DanielYK: So does this endpoint have an issue with other IVRs? |
| 14:10.02 | Samot | Or just this specific one? |
| 14:10.55 | DanielYK | At the moment I am only aware of one problem with exactly this one. I have tested some other callcenter numbers. |
| 14:11.30 | DanielYK | The setup was in prod use without problems since today, where someone called this new IVR number. |
| 14:11.45 | sibiria | file: right, i wouldn't suggest for that to be enforced, but available as an option, to avoid having to go the dial plan path we do now |
| 14:12.01 | DanielYK | *until today |
| 14:12.06 | Samot | Do you have other remote endpoints? |
| 14:12.16 | DanielYK | No |
| 14:12.37 | Samot | Well it's one destination.... |
| 14:12.49 | Samot | So that's going to be hard to troubleshoot. |
| 14:13.05 | Samot | Because not only it is one destination but it's only this endpoint that has an issue with it. |
| 14:13.16 | Samot | Which is pointing the finger at the Fritz!Box. |
| 14:13.51 | DanielYK | I can test it with a remote softphone, one moment. |
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| 14:16.21 | DanielYK | The softphone has also the same problem with outgoing calls to this specific IVR. Other IVRs works without problem, also incoming calls are fine. |
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| 14:22.17 | Samot | Then you have to look at those individual calls to that specific destination. |
| 14:22.35 | Samot | And then compare them to other calls from local endpoints that seem to be able to call that destination without issue. |
| 14:22.40 | Samot | What is the difference? |
| 14:23.56 | DanielYK | rtp_keepalive for the endpoint fixed the issue. To I hear the keepalive packages? |
| 14:23.59 | DanielYK | *Do I |
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| 14:25.14 | Samot | For the Fritz!box endpoint? |
| 14:25.59 | DanielYK | Yes, the endpoint with the problem. |
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| 14:30.46 | DanielYK | What are these rtp keepalive packages? |
| 14:31.15 | Samot | What do you mean? |
| 14:32.07 | file | they're a CNG RTP packet if I recall, whether the remote side will do anything/care is up to them |
| 14:35.22 | DanielYK | Okay. Thank you all for the support. |
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| 15:23.49 | igcewieling | If anyone is on 2.11 this script will rename conference recordings so they show up in the GUI. https://pastebin.com/WqsaJk0F |
| 15:29.23 | igcewieling | heh, wrong channel |
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| 19:24.25 | Toerkeium | hello guys |
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| 21:54.26 | nny | tracking down iax2 issue with UNREACHABLE. Here's a pastebin from tcpdump, nmap and iax debug. https://pastebin.com/cwAPywq4 - Trying to diagnose further via any available tools has hit a dead end. Thoughts? |
| 22:00.54 | nny | Sorry here's a better set of pastebins including iax2 config Output from debug etc https://pastebin.com/bPewj3gt and output from iax2.conf https://pastebin.com/FSkeu3mm |
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