| Sicelo | she can install those two i suggested | 00:00 |
|---|---|---|
| Sicelo | maybe they'll give a clue | 00:00 |
| Juest | ok | 00:01 |
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| DocScrutinizer05 | stock SIP works just fine for me | 00:13 |
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| Juest | Well no idea, using Linphone.org and it doesnt work | 00:21 |
| Juest | i dont receive calls on the pc | 00:21 |
| * Sicelo has two SIP accounts (free) .. no one to call, haha | 00:24 | |
| Juest | And apparently i cant call to other networks because it asks authentication :| | 00:27 |
| Juest | Sicelo: linphone.org? | 00:29 |
| Sicelo | i have that too | 00:29 |
| Juest | Well, it does work but its sort of flacky :( | 00:30 |
| Sicelo | want to call? i've just put it online .. id is the same as my nick | 00:30 |
| * Juest feels dumb | 00:30 | |
| Juest | Okay, with jitsi i'll try | 00:30 |
| Juest | Connecting... | 00:31 |
| Juest | Nothing happens :s | 00:32 |
| Sicelo | hmm | 00:32 |
| Sicelo | sip:<nick>@sip.linphone.org? | 00:33 |
| Juest | Yes | 00:33 |
| Juest | and sip: is optional | 00:34 |
| Juest | or maybe not | 00:34 |
| Sicelo | on my N900, only ever used sip with my asterisk server (3 years ago) ... never really used this linphone account | 00:35 |
| Sicelo | pm me yours i'll try to call | 00:35 |
| Juest | Sicelo: Did a notice... lol | 00:35 |
| Juest | There it is | 00:36 |
| Juest | (i am using stereo mix for microphone) | 00:37 |
| Juest | i'll try now... | 00:37 |
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| Sicelo | lol .. my battery died | 00:37 |
| DocScrutinizer05 | (linphone) oh, on PC. well I tested all that was available, like 10 years ago, and finally ended at twinklephone which evolved to become the defacto reference implementation carriers use to test their own services against | 00:37 |
| Juest | Apparently i have issues making calls with Jitsi | 00:37 |
| Juest | as well as receiving calls via google talk | 00:37 |
| Juest | lol rip | 00:38 |
| Juest | Thank you anyway buddy | 00:38 |
| Sicelo | my battery clearly has too much internal resistance .. had 26% | 00:39 |
| Sicelo | Juest: so did you get a call at least? | 00:39 |
| DocScrutinizer05 | only downside of twinklephone: it makes thorough use of ALSA unique features which are not available under PukeAudio's crappy ALSA emu | 00:39 |
| Juest | Yes, i got a call but i cant call you | 00:39 |
| Juest | And my SGS4 GT-I9500 battery sucks | 00:40 |
| DocScrutinizer05 | when you can't call out, your registrar server is not working | 00:40 |
| Sicelo | how do you do it? i used the format above .. | 00:40 |
| Juest | i cant use the phone for long periods of time else it drops dead | 00:40 |
| Juest | I am putting just <nick>@sip.linphone.org | 00:40 |
| Juest | which is the correct way | 00:40 |
| Sicelo | Juest: yes, what DocScrutinizer05 is saying .. linpohone is actually very fussy even on N900 | 00:40 |
| Juest | sip: prefix is accepted too | 00:41 |
| Sicelo | check your configs :) | 00:41 |
| Juest | Tried ekiga which was worse | 00:41 |
| DocScrutinizer05 | sip.linphone.org, ojmy | 00:41 |
| Juest | Jitsi. | 00:41 |
| Juest | On Windows | 00:41 |
| Sicelo | i thought sip: was mandatory .. but then .. /me doesn't use SIP .. wish I would, but no one wants to use it | 00:41 |
| DocScrutinizer05 | there were several such crappy "community originated services", I recall ekiga which also never really worked | 00:41 |
| Juest | it is mandatory somewhere else, i am not sure where | 00:42 |
| Sicelo | yes, ekiga can't work on N900 due to some setting they need, and N900 can't be set for it | 00:42 |
| Juest | ekiga only supports a certain REGISTER approach | 00:42 |
| DocScrutinizer05 | I'm talking about the servers | 00:42 |
| DocScrutinizer05 | not the client | 00:43 |
| DocScrutinizer05 | you know, SIP oughta be SIP | 00:43 |
| DocScrutinizer05 | twinklephone can do it all | 00:43 |
| DocScrutinizer05 | it has a bazillion config options | 00:43 |
| Sicelo | DocScrutinizer05: that's a server? or service? :) | 00:44 |
| DocScrutinizer05 | to accomodate the last idiotic server | 00:44 |
| DocScrutinizer05 | twinklephone.org | 00:44 |
| DocScrutinizer05 | err .com | 00:44 |
| DocScrutinizer05 | http://www.twinklephone.com/ | 00:44 |
| Sicelo | ah, client | 00:45 |
| DocScrutinizer05 | I'm using it with https://www.sipgate.de/ service | 00:46 |
| DocScrutinizer05 | since like 10 years, always works | 00:46 |
| Juest | Oh well | 00:46 |
| Juest | So what SIP free service i should use/be using!? | 00:47 |
| * Juest hates himself | 00:47 | |
| Sicelo | Juest: if N900 can do it, then Jitsi should be able .. just recheck your settings. :) | 00:47 |
| Juest | its just stuck on Connecting | 00:47 |
| Juest | Auto DTMF, REGISTER keep-alive, no authorization name, support for encrypted calls enabled (zrtp), SIMPLE presence | 00:48 |
| Juest | These are the settings | 00:48 |
| DocScrutinizer05 | Juest: (which service) that clearly depends on where you're living | 00:48 |
| Juest | There's barely services.... | 00:49 |
| Juest | I am in argentina | 00:49 |
| Juest | Oksana's in Australia... | 00:49 |
| DocScrutinizer05 | also completely free services are very likely to be shitty, since nobody has something to give away for free | 00:49 |
| Juest | lel | 00:49 |
| Juest | most of them are european | 00:50 |
| Sicelo | transport - udp, outbound proxy - sip.linphone.org, port 5060, loose routing, REGISTER keep alive, | 00:50 |
| Juest | linphone uses TLS | 00:50 |
| DocScrutinizer05 | I guess there must be international commercial services available now that allow call-out for tiny money, free call-in, and of course free intra-service calls | 00:50 |
| DocScrutinizer05 | free world dialup comes to mind, dunno if that still exists | 00:51 |
| Sicelo | Juest: can you call others besides Oksana? also, our N900 doesn't do ZRTP. as for TLS .. dunno .. but what i showed above are the 'working' settings that are currently on my n900 | 00:51 |
| Juest | No clue about working ZRTP and sip.linphone.org is the registar | 00:51 |
| Juest | Dont have anyone else | 00:51 |
| DocScrutinizer05 | hehe, twinkle was the first client to implement ZRTP | 00:52 |
| Juest | Well, i do but different networks doesnt work well | 00:52 |
| Juest | from linphone official client it asks me Realm Authentication | 00:52 |
| DocScrutinizer05 | get twinkle | 00:52 |
| Juest | Yes, theres still dial-up DocScrutinizer05, but most phones dont have such function | 00:53 |
| DocScrutinizer05 | huh? | 00:53 |
| Juest | true dial-up which is phone call based | 00:53 |
| DocScrutinizer05 | you lost me | 00:53 |
| Juest | > free world dialup comes to mind, dunno if that still exists | 00:53 |
| * Sicelo drifts off to other things ... | 00:53 | |
| DocScrutinizer05 | "freworld-dialup" is a company: fwd or somesuch | 00:54 |
| Sicelo | let me know if you/Oksana ever find anything about the skype problem | 00:54 |
| Juest | Sure. | 00:54 |
| Juest | laters sicelo | 00:54 |
| Juest | have a good night man | 00:54 |
| Sicelo | i've ordered two supposedly original batteries .. getting them next week. i hope my N900 will get normal agin | 00:55 |
| Sicelo | *again | 00:55 |
| DocScrutinizer05 | http://www.freeworlddialup.com/ | 00:55 |
| DocScrutinizer05 | http://www.voip-info.org/wiki/view/Free+World+Dialup | 00:56 |
| DocScrutinizer05 | http://www.voip-info.org/wiki/view/Sip+Trunking+Providers | 00:58 |
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| * Juest checks out | 01:01 | |
| DocScrutinizer05 | and again, twinkle. Eventually Michel deBoer implemented a setup wizard for some VoIP provider who probably paid for that to offer a decent client to their customers. Might still work | 01:02 |
| DocScrutinizer05 | with twinkle on both ends, decent DynDNS or fixed IP, and a tad of NAT fiddling, you could even do direct IP2IP SIP, without any registrar | 01:04 |
| Juest | :p | 01:04 |
| DocScrutinizer05 | if by any chance you could use the german locale in twinkle, try it. It's *awesome* (and by me ;-P ) | 01:05 |
| DocScrutinizer05 | suse has twinkle1.9 | 01:06 |
| Juest | lol | 01:06 |
| DocScrutinizer05 | nfc what they did for twinkle 1.5 to 1.8 | 01:07 |
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| Juest | I guess i should disable encryption thing | 01:09 |
| DocScrutinizer05 | ZRTP? why? | 01:09 |
| Juest | Because thats probably making it get stuck on Connecting | 01:10 |
| DocScrutinizer05 | it should 'just work' and remain inactive as long as far end doesn'T support | 01:10 |
| DocScrutinizer05 | nope, ZRTP establishes connections absolutely transparent | 01:10 |
| Juest | "Indicate support of ZRTP in signaling protocol" | 01:10 |
| Juest | ^ a checkmark | 01:10 |
| DocScrutinizer05 | yes | 01:11 |
| DocScrutinizer05 | this is just a signal you're sending that your client can handle ZRTP | 01:11 |
| DocScrutinizer05 | doesn't cause any known problems | 01:11 |
| DocScrutinizer05 | but sure, disable it | 01:12 |
| DocScrutinizer05 | can't hurt | 01:12 |
| Juest | Then no idea why its stuck on "Connecting" when i tried to call | 01:12 |
| DocScrutinizer05 | ethereal is your friend | 01:12 |
| DocScrutinizer05 | cone NAT? | 01:12 |
| DocScrutinizer05 | enable STUN! | 01:12 |
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| Juest | About that... it works ok when calling over gtalk | 01:14 |
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| Juest | it fails to establish ICE when incoming call connects | 01:14 |
| Juest | wtf | 01:14 |
| DocScrutinizer05 | gtalk is XMPP iirc | 01:14 |
| DocScrutinizer05 | ICE ~= STUN | 01:14 |
| DocScrutinizer05 | I guess you have a nasty NAT | 01:15 |
| DocScrutinizer05 | open your port 5060 and the RTP ports your service recommends | 01:15 |
| Juest | I am on DMZ | 01:15 |
| DocScrutinizer05 | hmm | 01:15 |
| Juest | The issue is on the other side because.... well | 01:16 |
| Juest | its a default NAT'd network | 01:16 |
| DocScrutinizer05 | DMZ still doesn't mean your outbound port also creates an inbound path on same IP:port tuple | 01:16 |
| DocScrutinizer05 | particularly for UDP | 01:16 |
| Juest | When *she calls* is when the issue happens | 01:16 |
| Juest | not when i do | 01:16 |
| DocScrutinizer05 | *DIS*able ICE/STUN for testing | 01:16 |
| DocScrutinizer05 | sorry, I'm out for a beer | 01:17 |
| DocScrutinizer05 | ethereal is your friend | 01:17 |
| Juest | ICE, UPnP, Auto discover STUN/TURN, own STUN severs, Jingle Nodes | 01:18 |
| Juest | is what's available for gtalk in jitsi | 01:18 |
| DocScrutinizer05 | dunno why we're talking jitsi - never even heard of it | 01:19 |
| Juest | Because i am using that client | 01:20 |
| Juest | it was previously known as SIP Communicator or something | 01:20 |
| Juest | I am here because of Oksana and i am not a maemo user but trying to troubleshoot some maemo issues with her :) | 01:21 |
| DocScrutinizer05 | https://www.sipgate.de/basic/signup | 01:22 |
| DocScrutinizer05 | so when she's using maemo then why is jitsi relevant in all this? | 01:23 |
| Juest | Because i am the Jitsi for Windows user | 01:23 |
| Juest | ~_~ ffs | 01:25 |
| DocScrutinizer05 | and i'm a twinkle on linux user, still how's that relevant for Oksana's problem? | 01:28 |
| DocScrutinizer05 | your jitsi doesn't need any special settings to talk to maemo SIP client | 01:29 |
| Juest | Well | 01:31 |
| Juest | Linphone client didnt work | 01:31 |
| Sicelo | and we've partly 'proved' that maemo's SIP client is working fine .. deduction - oksana likely has no problrm either | 01:32 |
| * DocScrutinizer05 thinks it's extremely inefficient and silly to debug person C's problems by talking via proxy person B | 01:32 | |
| DocScrutinizer05 | even the statements are highly ambiguous - I don't have a clue who's using linphone, on which platform, and why | 01:34 |
| Juest | linphone has an official app | 01:34 |
| Juest | LOL | 01:34 |
| Juest | She tried to call me with me using that application | 01:34 |
| Juest | and it did not work | 01:34 |
| Juest | nothing happened on my side | 01:34 |
| DocScrutinizer05 | no, linphone app eventually got an 'oggicial' server which is supposed to be crap | 01:34 |
| DocScrutinizer05 | original even | 01:35 |
| DocScrutinizer05 | err official | 01:35 |
| Sicelo | Juest: maybe N900 enforces 'sip:' convention? | 01:35 |
| Juest | ... | 01:35 |
| DocScrutinizer05 | Sicelo: nope | 01:36 |
| Sicelo | ah, it appends it .. just tesed | 01:36 |
| Sicelo | *prepend, rather | 01:36 |
| Sicelo | (tested) | 01:36 |
| DocScrutinizer05 | check the RFCs listed on twinklephone.com | 01:36 |
| DocScrutinizer05 | maemo client is mostly compliant | 01:37 |
| DocScrutinizer05 | linphone (server / client) though, NFC if it's even SIP still | 01:37 |
| DocScrutinizer05 | tbh linphone never worked for me with sipgate registrar | 01:38 |
| Juest | Yeah, linphone clients asks me for Realm Authentication when i try to call to a user on a different registrar | 01:38 |
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| DocScrutinizer05 | first you need to know if the different registrar even allows dial-in | 01:39 |
| DocScrutinizer05 | most don't anymore | 01:40 |
| Juest | Even if it does it asks for login | 01:40 |
| Juest | i think | 01:40 |
| DocScrutinizer05 | quite possible, linphone isn't made to handle this scenario I guess | 01:40 |
| DocScrutinizer05 | btw neither is maemo telepathy SIP client | 01:41 |
| DocScrutinizer05 | the vast majority of SIP clients only knows INVITE via registar server as proxy | 01:41 |
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| DocScrutinizer05 | http://www.voip-info.org/wiki/view/SIP | 01:46 |
| DocScrutinizer05 | https://en.wikipedia.org/wiki/Session_Initiation_Protocol | 01:49 |
| Juest | :|} | 01:49 |
| DocScrutinizer05 | particularly https://en.wikipedia.org/wiki/File:SIP_call_flow_between_UA,_Redirect_Server,_Proxy_and_UA.png | 01:50 |
| DocScrutinizer05 | ((SIP clients only knows INVITE via registar server as proxy)) directly calling the callee's registrar proxy would be directIP2IP SIP which most clients can't do. Sending an INVITE of a different registrar's client to your own registrar is often not working since most dervices have no interest to support calls to other services | 01:57 |
| DocScrutinizer05 | s/dervices/services/ | 01:58 |
| infobot | DocScrutinizer05 meant: ((SIP clients only knows INVITE via registar server as proxy)) directly calling the callee's registrar proxy would be directIP2IP SIP which most clients can't do. Sending an INVITE of a different registrar's client to your own registrar is often not worki... | 01:58 |
| DocScrutinizer05 | ~useless | 01:58 |
| * infobot starts crying and hides from docscrutinizer05 in the darkest corner of the room. :( | 01:58 | |
| DocScrutinizer05 | neither do they usually support inbound SIP calls from foreign origin | 01:59 |
| DocScrutinizer05 | the result of doing such inbound SIP call nevertheless is the far end registrar probably answering a by a 401 UNAUTHORIZED which causes caller's UA to pop up a registration dialog to enter the credentials for that assumed account | 02:01 |
| DocScrutinizer05 | note that the INVITE you send to *your* registrar (so it might forward it to the destination) is identical to the INVITE to the foreign registrar if your registrar forwards it. So the foreign registrar will think it's one of their clients who "forgot to REGISTER" | 02:04 |
| Oksana | Sicelo, DocScrutinizer: Thanks a lot! As the very least, linphone now manages to sign in (previously got Network Error, tried different settings randomly; Sicelo's settings just work) | 02:04 |
| DocScrutinizer05 | Oksana: if you want a decent SIP gateway, feel free to query me for cheating the data on sipgate.de registration | 02:05 |
| DocScrutinizer05 | https://www.sipgate.de/basic/signup | 02:06 |
| DocScrutinizer05 | it's a trunking service only | 02:06 |
| DocScrutinizer05 | means it does SIP only to clients | 02:07 |
| DocScrutinizer05 | err customers | 02:07 |
| DocScrutinizer05 | it allows free landline call-in though | 02:07 |
| DocScrutinizer05 | and of course free SIP calls between customers | 02:07 |
| DocScrutinizer05 | they once offered SIP dial-in too, but this feature got cancelled worldwide basically (all trunking providers) since spammers used that feature for unsolicited calls to victims | 02:09 |
| DocScrutinizer05 | maybe you find a better service provider in one of the tutorials sites I linked | 02:11 |
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| Juest | Ok | 02:12 |
| DocScrutinizer05 | https://www.google.de/search?q=free+SIP+network+service+provider | 02:14 |
| DocScrutinizer05 | http://www.voip-info.org/wiki/view/Free+VoIP+Networks | 02:15 |
| DocScrutinizer05 | please forget linphone provider/registrar, it sounds just odd and prone to introduce problems of all sorts | 02:19 |
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| Juest | Oh dear | 02:20 |
| DocScrutinizer05 | I don't think you shpould call me 'dear', Oksana may though ;-) | 02:22 |
| Juest | Lol | 02:22 |
| Juest | Oksana: ping | 02:23 |
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| Oksana | pong. And will try out sipgate | 02:40 |
| Oksana | "Oh dear" is just a saying, I guess, not literal ;-) | 02:43 |
| Juest | :) | 02:47 |
| Juest | yes | 02:47 |
| Juest | Oksana: And do try logging too please | 02:47 |
| Juest | Oksana: apparently got told that skype issue is codec crash | 02:48 |
| Oksana | Do I understand correctly that sipgate requires a street address, and one in Germany at that?.. | 02:49 |
| * Oksana thinks that it is very nice of them to ensure proper emergency call functionality | 02:49 | |
| Oksana | Though the website does resemble a maze, somewhat | 02:52 |
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| * Oksana goes back to https://www.sipgate.de/basic/ | 02:53 | |
| Juest | o_O? | 02:55 |
| * Juest is confused | 02:55 | |
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| Juest | How i can do a P2P SIP call? | 03:14 |
| Juest | ^ Oksana asked this btw | 03:14 |
| gry | useername@ip instead of username@SIP.host | 03:15 |
| Juest | I mean, SIP usually requires an account | 03:15 |
| Juest | in maemo, how to call without a sip account | 03:16 |
| Oksana | Juest: That would need an SIP server installed on the ip. And, I do not see whether there are any SIP servers available for Maemo 5, yet | 03:16 |
| Juest | Not that | 03:17 |
| Oksana | sipwitch sounds interesting | 03:17 |
| Juest | :| | 03:17 |
| Oksana | SIP (Session Initiation Protocol) is in principle a client-server protocol. It is possible for an endpoint of a communication session to implement both a user-agent server and a user-agent client, without the mediation a central switching system. https://en.wikipedia.org/wiki/Peer-to-peer_SIP | 03:19 |
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| DocScrutinizer05 | ((in maemo, how to call without a sip account)) toldya the telepathy client can't do that | 04:01 |
| Juest | damnit, why always when oksana is "away" | 04:03 |
| Juest | maemo google talk doesnt seem to have working video | 04:10 |
| Juest | telepathy* | 04:11 |
| Juest | i am unable to see video and oksana is unable too | 04:11 |
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| Juest | Video Call with SIP does not work | 05:27 |
| Juest | its either stuck calling or it dies if attempted from a ongoing call | 05:28 |
| gry | call a test dummy robot instead of calling someone on wifi :p | 05:29 |
| gry | ekiga used to have 500@ekiga.net for tests, but maybe it only takes tests from ekiga customers, and you have to find another one for your server, or find a robot account which takes random guest calls | 05:29 |
| gry | a tox client for maemo could also be kinda cool | 05:30 |
| * gry has 4 app ideas already, with zero ui knowledge on this platform ... | 05:30 | |
| Oksana | Yes, I remember. Help (aka gui for man, or something), calculator-ui (in Gtk, not in Qt4), calculator engine (if I see it correctly, currently calculator engine is closed-source?), sketch (again, Nokia's Sketch seems to be closed source), and now tox plugin for telepathy?.. | 05:33 |
| Juest | And still would like to troubleshoot our communication issues | 05:34 |
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| gry | Turns out that Firefox instance crashed while doing a Windows upgrade. Darn. I better get my notes together and get some things done. | 06:07 |
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| Juest | :| | 06:09 |
| Oksana | :? | 06:10 |
| Oksana | By the way, am I the only one having issues with imgur , does it work for you, or does it not-work for you? | 06:11 |
| Juest | Works fine | 06:15 |
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| Juest | gry now it doesnt let me | 09:08 |
| Juest | return | 09:08 |
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| KotCzarny | vajb: yes, there is /tmp/ file that you can write commands | 09:45 |
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| Juest | Apparently Voice Calls work properly, so does IM | 23:56 |
| Juest | but not SIP presence and calls stop working as soon video is attempted | 23:57 |
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