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| 01:42.19 | *** join/#asterisk Casper (~Casper@unaffiliated/thephantom) |
| 01:43.16 | Casper | hi there, does anyone have some good inexpensive voip provider to suggest? the one I currently use is going downhill with audio issues... |
| 01:43.29 | Casper | with canadian 911 support of course |
| 02:01.02 | Samot | It would probably help to know the current one. |
| 02:01.15 | Casper | voip.ms |
| 02:02.34 | Casper | I had occasional issues in the past, but usually it was some network issues they had or alike, but this year... it's quite annoying... no outgoing audio... |
| 02:03.06 | Samot | Well I haven't seen a lot of reports like that for them. |
| 02:04.22 | Samot | When did this no outbound audio issue start? |
| 02:05.26 | Casper | early this year |
| 02:05.32 | Casper | it's random |
| 02:06.20 | Casper | I even changed my ATA, but today it did it on a forward to cellphone, so it prove that it is not my isp, net, router or ata |
| 02:08.27 | Samot | OK so you have random outbound audio from them. What have they said about this? |
| 02:08.42 | Samot | Because if this was really a wide spread issue, it would have been commented on widely. |
| 02:09.01 | Samot | Not just wide spread but a problem that has existed for the better part of the year. |
| 02:09.08 | Casper | no issue reported on their end, and they don't see anything wrong with my setup |
| 02:09.18 | Samot | Casper: What is your upload speed on your Internet connection? |
| 02:10.15 | Casper | but their clients ain't reporting stuff in review... few years ago they went very unstable (ddos, hardware issues on the provider, capacity issue) |
| 02:10.41 | Casper | Samot: 400/50, unloaded during the last test I did with them |
| 02:11.27 | Casper | but again, today's issue did not touched my network at all |
| 02:12.23 | Samot | And do you have audio issues during inbound calls when you are experiencing issues with outbound calls? |
| 02:12.44 | Casper | inbound appear fine |
| 02:13.06 | Samot | So during a time you have issues with outbound audio, your inbound audio is 100% fine? |
| 02:13.37 | Casper | as far as I know, yes |
| 02:14.03 | Samot | OK, so who is your ISP? |
| 02:15.16 | Casper | videotron |
| 02:15.22 | Samot | OK so a reseller. |
| 02:15.28 | Casper | no |
| 02:15.32 | Samot | Yes. |
| 02:15.43 | Samot | Does videotron own the infrastructure? |
| 02:15.44 | Casper | videotron is THE cable provider here, not a reseller |
| 02:15.45 | Casper | yup |
| 02:15.55 | Samot | OK. |
| 02:16.01 | Samot | So here's the thing... |
| 02:16.24 | Samot | Outbound is rarely a problem. It's generally inbound due to NAT, routers, networking, etc. |
| 02:16.38 | Casper | sometime the incomming cut mid call |
| 02:17.07 | Samot | What do you mean by that? |
| 02:17.11 | Samot | The audio is lost? |
| 02:17.20 | Samot | But comes back or the call drops? |
| 02:18.00 | Casper | never come back, have to hang up and call back |
| 02:18.17 | Samot | But this is on an incoming call or an outbound call? |
| 02:18.18 | Casper | sometime the no incomming audio issue stay. |
| 02:18.21 | Casper | both |
| 02:18.36 | Casper | usually the incomming however |
| 02:18.42 | Samot | Well that's more a NAT/networking issue. |
| 02:18.46 | Casper | including calls from the same provider |
| 02:18.55 | Samot | I understand that. |
| 02:19.14 | Samot | You're describing an issue that is NAT/networking related. |
| 02:19.18 | Casper | but again, today it was a call forward by voip.ms directly, the call did not pass throught my setup |
| 02:19.48 | Samot | So they forwarded a call and there was no audio? |
| 02:20.08 | Casper | exactly |
| 02:20.16 | Samot | For which side? |
| 02:21.37 | Casper | bro's cellphone -> voip.ms -> ring group, answered by the -> call forward to cellphone -> dropped incomming audio. Then he called back and no incomming audio, third call it was fine... |
| 02:22.37 | Samot | Wait, a ring group? |
| 02:23.16 | Casper | yeah, it ring the home phone + cellphone, first one that pick the line get the call |
| 02:23.27 | Casper | home being voip/ata |
| 02:23.49 | Samot | OK so you don't answer either of those and it forwards to where? |
| 02:26.19 | Casper | DID -> ring group, no answer to voicemail, ring group hold the ATA and the cellphone. |
| 02:26.46 | Samot | OK so this isn't forwarding. |
| 02:27.00 | Samot | This is a ring group making a call and you pickup on your cell phone |
| 02:28.45 | Casper | forwarding to cellphone (not voip, true cell) |
| 02:30.47 | Samot | It's not really forwarding. |
| 02:31.00 | Samot | The call hits a ring group and it rings two destinations. |
| 02:31.11 | Samot | You could answer on your ATA or your cellphone. |
| 02:31.21 | Samot | If no one answers it goes to voicemail. |
| 02:31.32 | Samot | So have you tried this with a direct forwarding? |
| 02:31.56 | Samot | No ring group, no ringing your ATA. Just sending the call straight to your cell phone. |
| 02:31.57 | Casper | no, the issue fixed itself by the time I was home, so can't do more testing |
| 02:32.14 | Casper | and that's the issue, it's intermittant |
| 02:34.39 | Samot | Well you're not going to find another inexpensive provider that does what VoIP.ms does most likely.. |
| 02:34.53 | Samot | Offering PBX features to home users isn't really a standard business model. |
| 02:43.08 | Casper | really, if I can get ring group, call forward and SIP it's fine |
| 02:43.23 | Casper | but yeah... voip is quite featurefull |
| 02:43.56 | Casper | also, inexpensive |
| 02:44.12 | Samot | Ring Groups are a PBX feature. |
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| 02:50.39 | Casper | I think I'll kill everything and redo all the config... I wonder if there is a corruption on the config on their side that could be an issue... |
| 02:51.07 | Casper | if I do, I kill the subaccount for the ata, factory default the router and ata and start over... |
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| 11:53.16 | Pinchiukas | I'm pretty new to Asterisk and I'm trying to run a PoC instance in a new environment. Can I somehow just allow any client to REGISTER? |
| 12:11.03 | Samot | Yes |
| 12:16.29 | Pinchiukas | Samot: maybe there is a quick setting I can flick for this to be possible? :) |
| 12:23.54 | Samot | What do you mean? |
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| 12:33.40 | Pinchiukas | Samot: I mean, how exactly do I achieve making Asterisk accept any registration from anywhere? I suppose I have to add something to some config file? |
| 12:34.55 | Samot | It's automatic. |
| 12:35.07 | Samot | Nothing stops registreations from other devices until you put something there. |
| 12:35.19 | Samot | So you create your endpoint/peer and you try to register. |
| 12:35.30 | Samot | Are you having issues registering a device? |
| 12:36.06 | Pinchiukas | I mean I just did a 'apt-get install asterisk' on a Ubuntu container and them I'm trying to connect to it from a softphone and I'm getting a 403. |
| 12:39.52 | Samot | So you didn't even set anything up. |
| 12:39.54 | sibiria | well you haven't added that user to your asterisk config, have you? |
| 12:40.03 | Samot | There's no default config for Asterisk. |
| 12:40.05 | Pinchiukas | Yeah, nothing. |
| 12:40.15 | Samot | Then you have a few things to do. |
| 12:40.34 | sibiria | sounds like you want to register "as anyone without password", not "from anywhere"? |
| 12:40.45 | Samot | But if you didn't create any accounts, how did you think you could register a device? |
| 12:40.46 | Pinchiukas | What's the complete minimum? I'm going to trash this instance in a few minutes so don't really care about security. |
| 12:40.57 | Pinchiukas | Samot: I'm pretty SIP-ignorant. :) |
| 12:40.58 | Samot | You don't need to trash the instance. |
| 12:41.06 | Samot | You need to config Asterisk. |
| 12:42.01 | Pinchiukas | I meant I don't care if the setup is insecure. |
| 12:44.51 | Samot | You need to config Asterisk. |
| 12:44.52 | Pinchiukas | sibiria: well probably both if there are limitations for where you can connect from. |
| 12:45.27 | Pinchiukas | I take it there is no easy way to make anyone able to just REGISTER without each client having to be explicitly configured? |
| 12:45.53 | Samot | OK. There is more to config than just the endpoints. |
| 12:46.03 | Samot | Have you read the wiki or the sample files? |
| 12:54.43 | Pinchiukas | I take it I'll have to spend several weeks reading documentation just to get an example call running? |
| 13:00.18 | Samot | No. |
| 13:02.35 | Samot | But one would figure that if you are SIP ignorant, as you pointed out, that you would have spent some time reading/learning about SIP and since your choice was Asterisk. The same thing there. |
| 13:07.42 | Pinchiukas | Is there a better tutorial to get me started quickly than reading the tonnes of documentation in the wiki? |
| 13:10.10 | Samot | I'm sure the Internet has something. |
| 13:11.08 | Pinchiukas | Ah cool. |
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| 14:45.07 | Zombie | You ever get the feeling your hard work is paying off? |
| 14:45.47 | Zombie | Samot: The Hospital I work for has approved training me on Voice and Data Techs for the Phones. |
| 14:48.02 | Samot | Oh boy. |
| 14:50.18 | drmessano | <PROTECTED> |
| 14:53.34 | Samot | Yeah. |
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| 16:08.17 | Pinchiukas | Ok I managed to set up a couple of users and perform a call. Is there a simple way to make Asterisk proxy all the media? |
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| 16:08.34 | Alex69 | Hey guys! |
| 16:19.30 | Alex69 | I've been stuck with configuring my asterisk server with openvpn. Second day, tried alot of stuff and still didn't figure it out. I did get it working on a Lan network, but that is not enough for my usecase. Basically I have an asterisk server setup on a private network (192.168.50.0/24). On the same network I do have an openvpn server setup, this |
| 16:19.30 | Alex69 | is a seperate server. Now I'm trying to setup a connection between a softphone on the 192.168.50.0network and a softphone outside of this network connected through a vpn. The openvpn server is publicly available so the second client connects to the openvpn over the internet to get access to the private server. The openvpn server is configured to |
| 16:19.31 | Alex69 | NAT, thus the ip packets that arrive on the private network through the vpn get translated to a private ip(192.168.50.60). When I tried this configuration without configuring any NATing in asterisk, the connection between the clients got setup but I only had one way audio (outside client audio passed to the private network, not the other way). I |
| 16:19.31 | Alex69 | started looking around and saw that people are configuring NATing in sip.conf, I did the same stuff. It didn't seem to do anything and the issue persisted. Everytime I changed the config, I restarted the asterisk server to be sure that the config passed on. While testing I noticed that suddenly all the audio dissapeared. I restored the sip.conf I |
| 16:19.32 | Alex69 | had in the beginning but the audio didn't return for some reason, I'm kind of struck why this would happen. Is there some config cached somewhere? After googling some extra hours I realised that sip.conf is the deprecated stuff and pjsip.conf is the new guy. Since I had my routes setup using pjsip.conf it sounded like a good idea to configure the |
| 16:19.32 | Alex69 | NATing there, which didn't give the required result. I'm stuck and I hope anyone in here can point me in the right direction. Basically I need to setup a connection with audio between two softphones, where one softphone is connecting to the private network over an openvpn. As stated before, the openvpn is configured using NATing and not Routing. |
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| 18:15.54 | igcewieling | turn off direct media |
| 18:16.52 | Pinchiukas | Yeah looks like I've done that kind of successfully. But it appears that the Telephone app on Mac chooses next port to the one that's given to it and media only travels one-way. |
| 18:17.03 | Pinchiukas | This is pretty weird... |
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| 23:07.14 | volga629 | Hello Everyone, asterisk JABBER_STATUS always report status 6 for any JID. Do I need some quotes around JID ? |
| 23:07.52 | volga629 | dial plan example https://paste.centos.org/view/d97f54dd |
| 23:12.05 | igcewieling | sorry, I don't use Jabber with Asterisk. |
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