openjarbas / baresipy Goto Github PK

You provide 2 example scripts, neither of which work (using either baresip 1.0.0 or 1.1.0 on Ubunto 20.04)

./spam_call.py
15:15:23.190 - baresipy:_handle_no_accounts:251 - DEBUG - No accounts setup
15:15:23.190 - baresipy:login:64 - INFO - Adding account: +442083XXXXXX
15:15:23.243 - baresipy:handle_ready:262 - INFO - Ready for instructions

And then nothing. No call placed,

./joke_bot.py
15:19:45.951 - baresipy:_handle_no_accounts:251 - DEBUG - No accounts setup
15:19:45.951 - baresipy:login:64 - INFO - Adding account: +2083XXXXXX
15:19:46.004 - baresipy:handle_ready:262 - INFO - Ready for instructions

And then nothing. No calls answered.

Hi,

Is there a way to initiate a videocall?

I am getting this unknown error-2 while running baresip script on auto run...But without autorun it is working fine..
PROBLEM -->> Cannot make outgoing call from Raspberry Pi . Baresip crashes when I am picking up the phone on incoming call
call state is Ringing after picking up phone It shows unknown error-2.....

10:06:04.838 - baresipy:run:370 - DEBUG - audio: Set audio decoder: opus 48000Hz 2ch
10:06:04.925 - baresipy:run:370 - DEBUG - audio: player started with sample format S16LE
10:06:04.973 - baresipy:run:370 - DEBUG - audio: Set audio encoder: opus 48000Hz 2ch
10:06:05.072 - baresipy:run:370 - DEBUG - warning: Unknown speex_preprocess_ctl request: 2
10:06:05.125 - baresipy:run:370 - DEBUG - warning: Unknown speex_preprocess_ctl request: 46
10:06:05.270 - baresipy:run:370 - DEBUG - warning: The VAD has been replaced by a hack pending a complete rewrite
10:06:05.470 - baresipy:run:370 - DEBUG - speex_pp: Speex preprocessor loaded: srate = 48000Hz
10:06:05.513 - baresipy:run:370 - DEBUG - �[31malsa: could not open ausrc device 'default' (No such file or directory)
10:06:05.566 - baresipy:run:370 - DEBUG - �[;m�[31maudio: start_source failed (alsa.default): Unknown error -2
10:06:05.579 - baresipy:run:370 - DEBUG - �[;msip:[email protected]: session closed: Unknown error -2
10:06:05.637 - baresipy:handle_call_status:314 - DEBUG - Call Status: DISCONNECTED
10:06:05.649 - baresipy:handle_call_ended:328 - INFO - Call ended
10:06:05.820 - baresipy:handle_call_ended:329 - DEBUG - Reason: Unknown error -2

Does this support direct url sip, and if yes how?

Hi @JarbasAl

I have a question, i see there is a transfer option, but i think this transfer option is only avaible when the call has been answered
What i want:
I have 2 accounts registered with baresip on a PBX
When there is an incoming call to user1 , can i immediately forward the call to user2 without user 1 thats answers the call?
I dont think i can do it with baresip, but maybe i can do it with your script?

thnx in advance

Tried to call to Linphone, got error in log:
18:31:56.752 - baresipy:run:289 - DEBUG - Creating UA for sip:adushkin1:**********@sip.linphone.org ...
18:31:56.754 - baresipy:run:289 - DEBUG - account: username:password is now disabled please use ;auth_pass=xxx instead

From Log:

CODE:
from baresipy import BareSIP
from time import sleep

to = "[email protected]"

gateway = "sip.linphone.org"
user = "adushkin1"
pswd = "*********"

class JokeBOT(BareSIP):
def handle_incoming_call(self, number):
self.accept_call()

def handle_call_established(self):
    self.speak("Welcome to the jokes bot")
    self.speak(get_joke())
    self.speak("Goodbye")
    self.hang()

b = JokeBOT(user, pswd, gateway, None, True)

b.call(to)

while b.running:
sleep(0.5)
if b.call_established:
b.send_dtmf("123")
b.speak("this is jarbas personal assistant speaking. this was a test")
b.speak("Goodbye")
b.hang()
b.quit()

I'm not sure what circumstances caused this file not to exist on my system, but the module does not handle this exception

I have a fonial voip, and I want to make calls on phone numbers