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| 01:08.44 | Samot | Well as much as I've tried to help this project and beta test..... |
| 01:09.29 | Samot | Just wow, take away actual technical issues the onboarding process is just not....pleasant. |
| 01:11.07 | Samot | The wizard takes you through, tells you that you're getting a number but you don't get to choose and you aren't told what is is. |
| 01:11.42 | Samot | It tells you to use a GS Wave and then gives you a QRCode for your creds. Except everything is jammed into four lines with no details on what is what. |
| 01:12.16 | Samot | Two SIP URIs but no indication what they are for, which one is the host/domain, if one is the OB proxy, etc... |
| 01:13.13 | Samot | It forces the extension selection. You don't get a choice in any of that. |
| 01:13.40 | Samot | But the best so far. You can't manage your account or SEE what DID was assigned to you until you register your first extension/account. |
| 01:14.22 | Samot | So thoughts on that on-boarding process? |
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| 03:01.11 | ruben23 | hello guys i have problem my VoIp carrier is not registering to my asterisk just getting this error ----> NOTICE[8060]: chan_sip.c:15906 sip_reg_timeout: -- Registration for 'vicidial1@xx.xxx.xxx.xxx' timed out, trying again (Attempt #6 |
| 03:01.49 | Samot | You mean you can't register TO your provider. |
| 03:01.49 | ruben23 | ov500/user011 xxx.xxx.xxx.xxx Yes Yes 5060 UNREACHABLE |
| 03:01.55 | Samot | It timed out |
| 03:02.09 | Samot | Which means it is not getting a response back from the other side. |
| 03:02.24 | ruben23 | Samot: on the carrier side.? |
| 03:02.47 | Samot | Is that who you are sending the register to? |
| 03:03.08 | Samot | Are they the other side? |
| 03:04.03 | drmessano | heh vicidial |
| 03:04.13 | ruben23 | Samot: yes - using username/password but with my other asterisk server its working |
| 03:06.15 | ruben23 | <PROTECTED> |
| 03:07.18 | Samot | OK well this one is having an issue where it is timing out. |
| 03:08.24 | Samot | drmessano: My kids are on their PS4's playing Minecraft. |
| 03:08.57 | Samot | drmessano: They don't believe that A) I have a friend that plays Minecraft and let alone, B) Has their own server. |
| 03:10.41 | drmessano | lol |
| 03:10.47 | drmessano | Well |
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| 03:11.01 | drmessano | Tell them that they're lame for playing on a console |
| 03:11.22 | Samot | lol |
| 03:11.35 | drmessano | I have a modded server and I play java, not lamer console |
| 03:12.45 | drmessano | dude, I got slams |
| 03:14.02 | Samot | hahaha |
| 03:14.15 | Samot | My one kid "Yeah, well I used to play on my PC then it died." |
| 03:14.44 | Samot | And something about it not playing right on his macbook |
| 03:14.51 | ruben23 | Samot: how to we correct timing out issue.? |
| 03:15.04 | Samot | Well, first you find out why there is one. |
| 03:15.18 | Samot | Are you using a domain name or an IP to connect to them? |
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| 03:50.22 | Slade | aww. flowsms went up in pricing.. |
| 03:59.33 | ruben23 | Samot: its IP based |
| 03:59.57 | Samot | OK, so you tried to at least ping it? Right? |
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| 04:56.28 | drmessano | lol |
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| 19:42.25 | anonymouz666 | is it possible to change the contact header to host instead of IP address? |
| 19:42.27 | anonymouz666 | PJSIP |
| 19:42.31 | file | no |
| 19:42.56 | anonymouz666 | not good :-( |
| 19:43.19 | anonymouz666 | this is needed |
| 19:43.30 | igcewieling | Use a SIP proxy then. |
| 19:44.09 | anonymouz666 | SIP proxy is just... boring! |
| 19:52.27 | Samot | Pardon? |
| 19:52.48 | seanbright | he said: SIP proxy is boring |
| 19:52.57 | seanbright | (or she) |
| 19:54.05 | Samot | anonymouz666: Please elaborate as to why a SIP proxy is boring. |
| 19:56.21 | anonymouz666 | I won't waste your time |
| 19:56.29 | drmessano | Thank you for that |
| 19:56.37 | seanbright | heh |
| 19:56.38 | Samot | Lol |
| 19:56.45 | seanbright | smart |
| 19:56.46 | seanbright | very smart |
| 19:56.55 | drmessano | Some heroes don't wear capes |
| 19:57.20 | Samot | And Asterisk doesn't do what is required it seems. |
| 19:57.54 | anonymouz666 | Some providers are inspecting the Contact to see if contains a FQDN or IP address. |
| 19:58.48 | file | Microsoft Teams requires a FQDN, but noone has contributed such functionality to set it |
| 19:58.50 | anonymouz666 | the session is entire over TLS, so you need to work with names. |
| 19:58.59 | anonymouz666 | that's it. |
| 19:59.13 | seanbright | no one uses teams |
| 19:59.22 | anonymouz666 | file: 1 jun will stop to work. |
| 19:59.32 | anonymouz666 | no more IP address. |
| 19:59.54 | anonymouz666 | seanbright: OpenSource is GREAT, but belieave in me, there millions using it |
| 20:00.09 | seanbright | i know, i was kidding |
| 20:01.10 | anonymouz666 | If we support the FQDN, no more vendors like AUC or Ribbon will take our space |
| 20:01.36 | seanbright | i'm sure there is reason we don't support it |
| 20:02.06 | seanbright | or i guess i should say, doing it just be non-trivial, otherwise sangoma would have already done it |
| 20:02.33 | Samot | Then again, this would be a Proxy/SBC job something Sangoma also does. |
| 20:03.00 | anonymouz666 | Sangoma uses FS on their SBCs |
| 20:03.40 | anonymouz666 | it is possible to change the outgoing contact-host for an INVITE |
| 20:03.45 | Samot | Yes, they use FreeSWITCH. With "switch" being the operable word. |
| 20:03.54 | anonymouz666 | but not answer 200 OK with the FQDN |
| 20:04.27 | igcewieling | Apparently most people don't need this feature or it would already be a feature. |
| 20:05.14 | seanbright | ^ this |
| 20:05.16 | Samot | Well... |
| 20:05.19 | anonymouz666 | MS is now asking for it, for sure someone will make it happen. |
| 20:05.28 | seanbright | i believe in you, anonymouz666 |
| 20:05.31 | seanbright | YOU CAN DO IT |
| 20:05.33 | anonymouz666 | FS already put the OPTIONS to include the host as MS DR needs it |
| 20:05.33 | Samot | In all fairness it tool 15 years for Asterisk to do multiple contact AORs. |
| 20:05.39 | Samot | took* |
| 20:06.09 | anonymouz666 | Asterisk took the wrong approach regarding to the SIP stack. Now, ex-post, is easy to say. |
| 20:07.29 | igcewieling | If you want access to the sip stack, then use a sip proxy or sbc. Asterisk is a PBX not a SIP router. |
| 20:08.00 | anonymouz666 | I know that since 2004, when Mark Spencer join my machine to fix a chan_agent bug. |
| 20:08.03 | anonymouz666 | lol |
| 20:09.41 | seanbright | i disagree that they took the wrong approach |
| 20:09.48 | seanbright | because they didn't use sofia? |
| 20:10.01 | seanbright | why do you feel it was 'the wrong approach?' |
| 20:10.13 | Samot | He won't waste your time. |
| 20:10.45 | anonymouz666 | chan_sip was too heavy to maintain to add feature or fix a bug without creating a bigger one |
| 20:12.52 | igcewieling | Sane people use pjsip now. |
| 20:13.38 | seanbright | you're saying that chan_sip was the wrong approach? |
| 20:13.41 | Samot | Oh, this is way better than answering why SIP proxies are boring. |
| 20:14.13 | seanbright | how many 1st class open source SIP stacks existed in 2002? (the oldest commit to chan_sip i can find in git) |
| 20:16.12 | joepublic | Setting up a PBX, "Asterisk is not a PBX, it is a voice toolkit" but regards sip routing, "Asterisk is a PBX"? |
| 20:17.26 | seanbright | huh? |
| 20:18.29 | igcewieling | "Asterisk is not a PBX, it is a toolkit which allows you to build a PBX." |
| 20:18.37 | joepublic | There we go :) |
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| 20:32.44 | jsmith | joepublic: with regards to SIP routing, Asterisk is a back-to-back user agent :-p |
| 20:36.12 | jsmith | seanbright: Exactly... If my aging memory still serves me, we had chan_sip, the sofia stack wasn't open source yet, and pjsip didn't exist in 2002. |
| 20:39.03 | arcarius | I tell people that Asterisk is to telephony as Apache is to web. It is a server that talks the protocols for you so you can build applications on top of it. |
| 20:54.02 | seanbright | so if i understand the goal, it is that when asterisk sends out an INVITE, the Contact header will contain a FQDN instead of an IP, correct? |
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| 20:55.05 | seanbright | file: would you do that in endpoint config or transport? |
| 20:55.15 | file | I'm not sure |
| 20:55.21 | seanbright | sure you are |
| 20:55.27 | file | it feels like a transport thing to me |
| 20:55.30 | seanbright | you have the whole thing worked out in your head |
| 20:55.33 | seanbright | i know |
| 20:55.44 | seanbright | but you're going to make me suffer through it |
| 20:55.47 | seanbright | this is the dynamic |
| 20:56.13 | file | statically dynamcizes seanbright |
| 20:58.53 | Samot | file: Perhaps more endpoint than transport. As with the endpoint you could shove the from_domain in there. |
| 20:59.15 | Samot | I'm only thinking that because a transport would support a single domain, would it not? |
| 20:59.17 | file | Contact controls routing though, which goes back to the transport, which is why I leaned toward that |
| 20:59.47 | Samot | Yeah, I get where you were coming from. I'm just seeing the whole "So now I gotta create 5 transports for 5 domains" |
| 20:59.48 | seanbright | i feel like external_signaling_address is the thing |
| 21:00.04 | Samot | What if everything is external? |
| 21:00.24 | seanbright | are shoes feet? |
| 21:00.42 | seanbright | i think i messed that up |
| 21:00.42 | Samot | Well I guess you could set the local_network to the /32 of the server... |
| 21:01.16 | seanbright | that would be a hack |
| 21:01.26 | Samot | Yes, it would be. No disagreement there. |
| 21:01.33 | seanbright | so something separate from the NAT handling stuff |
| 21:02.06 | seanbright | if someone wants to put together a kickstarter that'd be awesome |
| 21:02.18 | seanbright | shoot for i dunno... $25k? |
| 21:02.22 | seanbright | then i can get started! |
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| 22:33.18 | drmessano | 25k seems pretty trivial if he wants it |
| 22:33.42 | drmessano | Pretty sure the SIP thing is a paid feature, so someone must think having Asterisk work is worthwhile |
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| 22:36.36 | Samot | Yeah, plus you know $25K to get FQDN in the contact header to keep Ribbon Comm. out of your space of connecting to Microsoft Teams. |
| 22:36.45 | Samot | Well worth it. |
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| 22:53.21 | drmessano | lol |
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