s1p

I'd love to test, but I need to have a separate authenticating user than the sip account (number).
Do you have any plans implementing that in the near future?
Thanks a lot, shame on jolla for letting others do their work...
Maybe you'll on-board once telepathy-rakia and sofia-sip core packages will be finished? As high rated senior consultant :-)

Patiently Waiting Your Next Update on a TOUCH TONE feature to round up this Voip for me. This is legitimate. I can soon have a working phone line. VoipCheap intergrates with

S1P

and it works. ( speaker phone always echoed with this service, so having it start on speaker phone so i know the call connected is great. Next i click speaker phone off and put it to my face where there no longer an echo on the receivers end. Sometimes my face will hit the mute / headphone / speaker button because the proximity sensor does not shut the screen off, but im okay with manully tapping the power on my xa2 button everytime to ensure the screen is off before putting it up to my face. Thats probably phone specific programming or so something to fix/add, idk) But adding the KeyPress/Touchtone would make it perfect for me/some of us.

Wow, I was waiting for this since five years! :-) Thank you very much for your AWESOME work!

Did anybody test it with SIP-provider Easybell already? I can login successfully both via wifi and mobile data, but have some issues with outgoing calls (incoming calls not yet tested).

Bug list:

- Outgoing calls with SIP-provider Easybell do not work (all calls end immediately after dialling) (-> *1)
- Sorting order in contacts list does not use existing config of phone address book (s1p inevitably sorts by first name, phone address book sorts by users config, e.g. family name)
- Last action (e.g. "Call ended") is displayed until a new action is carried out. I guess, displaying the last message for only about 10 seconds should be enough. :-)

(*1) From log file: DEBUG [SIPServer-1] HandleResponseTerminatedOther [blabla@blabla] - call terminated with code: 400 session description parsing failed

Could you provide a Github-repo so we can better support you with bugs and logs? :-)

Again - THANK YOU for your great efforts!

OMG this thing works now. well, 1st, since i remembered my correct login in and password for my voip service, lol. I've been putting that in wrong probably all this time. next the VOIPCHEAP service that i use serve address is 77.72.169.134 with the 5060 port. Waiting a few hours after requesting SIP use via their/my settings online then checking back after a day clicking Enabled For SIP. It works, it really does. Been testing phone calls for the past three hours. ...mainly calling different businesses because i dont have any friends that i want to talk to. The top of my screen says 'Trying' and that is how i know it is working and logged in correctly to make outgoing calls. Im still testing it out but for the most part its good as its always sounded. Ive never used a different or more expensive provider which may work better in comparison, but considering how ive only paid 50 for service for close to a decade, im sticking with this company. A sad part is the touch tone key presses do not register at all so i cannot do any message trees or dial by extension type things (which is my perfered method of using the phone because who wants to talk to someone over the phone, not me)
Thanks for this sailfish os app that integrates with someones own VOIP provider! AND thanks for pushing out updates based on our feedback here. You deserve a coffee BTW where is the link to your paypal. I want to Send You COFFEE (two coffees if you get the Key Presses to register during calls)!

Problem solfed)))) ...next problem is the possibility to choose the speaker / mic devices. The menu has no response. Speaker is on by default and it's not possible to switch something.

Like many others, I'm very happy about a SIP-App too. But like normal for such a new app: I've got a problem and would like to report it. It's nothing critical so don't hesitate to ignore it if you don't have time or don't feel like taking a look at it.

I tried to use s1p as a client to a FritzBox 7560. Now I tried it with an internal call. I'm calling **620 (the internal number of s1p) from a DECT handset **610. Everything works fine (except that the speaker is on by default) which is really great work for such a new app. What doesn't work is hanging up. The audio connection is interrupted but the DECT handset doesn't receive the information that I hung up. Only when I hang up on the DECT handset too the connection is finally closed.

A log of the end of the connection is here:

[1] INFO handle_stdout - event frame: {'event': 'register_status', 'status': 19}
[1] INFO handle_event_frame - frame: {'event': 'register_status', 'status': 19}
[1] rec: b'{"event":"declined","from_peer":"christian","to_peer":"christian","outbound":true}\n'

[1] INFO handle_stdout - event frame: {'event': 'declined', 'from_peer': 'christian', 'to_peer': 'christian', 'outbound': True}
[1] INFO handle_event_frame - frame: {'event': 'declined', 'from_peer': 'christian', 'to_peer': 'christian', 'outbound': True}
b'PulseHandler - playback stream suspended - stream index: 0, suspended: false\n'
b'PulseHandler - unhandled message: &proto.Started{StreamIndex:0x0}\n'
[D] :86 - 1 handle_event_frame - unhandled event: declined
[1] INFO command_hangup - frame: {}
b'INFO [SIPServer-1] CommandHangup - hang-up call\n'
b'TRACE [SIPServer-1] HangupCall [9DBCA1B51AC44FC9@192.168.47.8], status: 17\n'
b'DEBUG [SIPServer-1] HangupCall [9DBCA1B51AC44FC9@192.168.47.8] - sending BYE\n'
[1] rec: b'{"event":"call_status","status":18}\n'

[1] INFO handle_stdout - event frame: {'event': 'call_status', 'status': 18}
[1] INFO handle_event_frame - frame: {'event': 'call_status', 'status': 18}
[1] rec: b'{"event":"call_bye","from_peer":"christian","to_peer":"**610","outbound":false}\n'

[1] INFO handle_stdout - event frame: {'event': 'call_bye', 'from_peer': 'christian', 'to_peer': '**610', 'outbound': False}
[1] INFO handle_event_frame - frame: {'event': 'call_bye', 'from_peer': 'christian', 'to_peer': '**610', 'outbound': False}
[1] rec: b'{"event":"result","command":"hangup","success":true}\n'

[1] INFO handle_stdout - event frame: {'event': 'result', 'command': 'hangup', 'success': True}
[1] INFO handle_event_frame - frame: {'event': 'result', 'command': 'hangup', 'success': True}
b'DEBUG [SIPServer-1] Bye [9DBCA1B51AC44FC9@192.168.47.8] - from: **610 (192.168.47.79:5060), ua: s1p, to: christian\n'
b'TRACE [SIPServer-1] UpdateCallStatus [9DBCA1B51AC44FC9@192.168.47.8] - updated status: 17 -> 18\n'
b'TRACE [SIPServer-1] WritePacket: 192.168.47.79:5060 => 192.168.47.8:5060 (BYE) sip:76A52E1F7FB9E00C9BBCC39384E96@192.168.47.8;transport=UDP\n'
b'BYE sip:76A52E1F7FB9E00C9BBCC39384E96@192.168.47.8;transport=UDP SIP/2.0\r\n'
b'Via: SIP/2.0/UDP 192.168.47.8:5060;branch=z9hG4bK830ADA8456674F32\r\n'
b'From: <sip:christian@fritz.box;transport=UDP>;tag=25642185\r\n'
b'To: <sip:**610@192.168.47.79:5060;transport=UDP>;tag=DDC4983E54787721\r\n'
b'Call-ID: 9DBCA1B51AC44FC9@192.168.47.8\r\n'
b'CSeq: 6 BYE\r\n'
b'Allow: INVITE, ACK, CANCEL, BYE\r\n'
b'Contact: <sip:christian@192.168.47.79:5060>\r\n'
b'Max-Forwards: 69\r\n'
b'User-Agent: s1p\r\n'
b'Content-Length: 0\r\n'
b'\r\n'
b'TRACE [SIPServer-1] HangupCall [3c571d3@192.168.47.79], status: 2\n'
b'TRACE [RTPServer-2] HandlePacket - playback stream not active\n'
b'TRACE [RTPServer-2] HandlePacket - playback stream not active\n'
[D] handle_event_call_status:139 - handle_event_call_status - updating status:  17 -> 18
[D] handle_event_call_status:156 - handle_event_call_status - setup:  false ringing: false answered: false hangup: true
b'TRACE [RTPServer-2] HandlePacket - playback stream not active\n'
b'TRACE [RTPServer-2] HandlePacket - playback stream not active\n'
b'TRACE [RTPServer-2] HandlePacket - playback stream not active\n'
b'TRACE [RTPServer-2] HandlePacket - playback stream not active\n'
b'TRACE [RTPServer-2] HandlePacket - playback stream not active\n'
b'TRACE [RTPServer-2] HandlePacket - playback stream not active\n'
b'TRACE [RTPServer-2] HandlePacket - playback stream not active\n'
b'TRACE [RTPServer-2] HandlePacket - playback stream not active\n'
b'TRACE [RTPServer-2] HandlePacket - playback stream not active\n'

That last message repeats till I hang up on my DECT handset.