Hi all,
I’ve been trying for a few days to simplify the pygui example to get rid of all tkinter gui. I’d like a simple agent to receive/place calls from CLI. This is my code
import pjsua2 as pj
import time
class Endpoint(pj.Endpoint):
"""
This is high level Python object inherited from pj.Endpoint
"""
instance = None
def __init__(self):
pj.Endpoint.__init__(self)
Endpoint.instance = self
def validateUri(uri):
return Endpoint.instance.utilVerifyUri(uri) == pj.PJ_SUCCESS
def validateSipUri(uri):
return Endpoint.instance.utilVerifySipUri(uri) == pj.PJ_SUCCESS
# Call class
class Call(pj.Call):
"""
High level Python Call object, derived from pjsua2's Call object.
"""
def __init__(self, acc, peer_uri='', call_id=pj.PJSUA_INVALID_ID):
pj.Call.__init__(self, acc, call_id)
self.acc = acc
self.peerUri = peer_uri
self.connected = False
self._onhold_ = False
def onCallState(self, prm):
ci = self.getInfo()
self.connected = ci.state == pj.PJSIP_INV_STATE_CONFIRMED
def onCallMediaState(self, prm):
ci = self.getInfo()
for mi in ci.media:
if mi.type == pj.PJMEDIA_TYPE_AUDIO and \
(mi.status == pj.PJSUA_CALL_MEDIA_ACTIVE or
mi.status == pj.PJSUA_CALL_MEDIA_REMOTE_HOLD):
m = self.getMedia(mi.index)
am = pj.AudioMedia.typecastFromMedia(m)
# connect ports
Endpoint.instance.audDevManager().getCaptureDevMedia().startTransmit(am)
am.startTransmit(Endpoint.instance.audDevManager().getPlaybackDevMedia())
if mi.status == pj.PJSUA_CALL_MEDIA_REMOTE_HOLD and not self.onhold:
print("'%s' sets call onhold" % self.peerUri)
self._onhold_ = True
elif mi.status == pj.PJSUA_CALL_MEDIA_ACTIVE and self.onhold:
print("'%s' sets call active" % self.peerUri)
self._onhold_ = False
def onDtmfDigit(self, prm):
# print("Got DTMF:" + prm.digit)
pass
def onCallMediaTransportState(self, prm):
# print("Media transport state")
pass
class Account(pj.Account):
def onRegState(self, prm):
print("***OnRegState: " + prm.reason)
def onMwiInfo(self, prm):
print("OnMwiState: " + prm.reason)
def onBuddyState(self, prm):
print("OnBuddyState: " + prm.reason)
def onIncommingSubscribe(self, prm):
print("OnSubscribeState: " + prm.reason)
def OnIncomingCall(self, prm):
c = Call(self, call_id=prm.callId)
call_prm = pj.CallOpParam()
call_prm.statusCode = 180
c.answer(call_prm)
ci = c.getInfo()
if input(f"Accept call from {ci.remoteUri}?") == u'yes':
call_prm.statusCode = 200
c.answer(call_prm)
else:
c.hangup(call_prm)
def initalise_sip_stack():
ep_cfg = pj.EpConfig()
ep = Endpoint()
ep.libCreate()
ep.libInit(ep_cfg)
sip_tp_config = pj.TransportConfig()
sip_tp_config.port = 5060
ep.transportCreate(pj.PJSIP_TRANSPORT_UDP, sip_tp_config)
ep.libStart()
acfg = pj.AccountConfig()
acfg.idUri = "sip:056004@172.20.1.8"
acfg.regConfig.registrarUri = "sip:172.20.1.8"
creds = pj.AuthCredInfo("digest", "*", "056004", 0, "alfatec")
acfg.sipConfig.authCreds.append(creds)
acc = Account()
acc.create(acfg)
return ep
if __name__ == "__main__":
endpoint = initalise_sip_stack()
time.sleep(600)
endpoint.libDestroy()However when I call from another contact (056003) to my agent … I can see there is no available account
15:30:50.045 pjsua_core.c .RX 949 bytes Request msg INVITE/cseq=11024 (rdata0x7fe8c4001c48) from UDP 172.20.1.8:5060:INVITE sip:056004@172.20.1.2:5060;ob SIP/2.0
Via: SIP/2.0/UDP 0.0.0.0:5060;rport;branch=z9hG4bKPj9599ad6d-2486-4483-8041-afac3bfdc05d
From: "MI 3" <sip:056003@172.20.1.8>;tag=e0d412a9-2d1d-45ef-be7f-04f0a8bdaf2e
To: <sip:056004@172.20.1.2;ob>
Contact: <sip:056003@0.0.0.0:5060>
Call-ID: 1b79c22a-dc7c-4652-92ae-416f3eafcd82
CSeq: 11024 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: "MI 3" <sip:056003@172.20.1.8>
Max-Forwards: 70
User-Agent: FPBX-15.0.23(16.16.1)
Content-Type: application/sdp
Content-Length: 233
v=0
o=- 1055573632 1055573632 IN IP4 172.20.1.8
s=Asterisk
c=IN IP4 172.20.1.8
t=0 0
m=audio 25808 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
--end msg--
15:30:50.045 pjsua_call.c .Incoming Request msg INVITE/cseq=11024 (rdata0x7fe8c4001c48)
15:30:50.045 pjsua_acc.c ..No available account to handle Request msg INVITE/cseq=11024 (rdata0x7fe8c4001c48)
15:30:50.045 pjsua_core.c ..TX 402 bytes Response msg 480/INVITE/cseq=11024 (tdta0x7fe8c4006b78) to UDP 172.20.1.8:5060:
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 0.0.0.0:5060;rport=5060;received=172.20.1.8;branch=z9hG4bKPj9599ad6d-2486-4483-8041-afac3bfdc05d
Call-ID: 1b79c22a-dc7c-4652-92ae-416f3eafcd82
From: "MI 3" <sip:056003@172.20.1.8>;tag=e0d412a9-2d1d-45ef-be7f-04f0a8bdaf2e
To: <sip:056004@172.20.1.2;ob>;tag=z9hG4bKPj9599ad6d-2486-4483-8041-afac3bfdc05d
CSeq: 11024 INVITE
Content-Length: 0
--end msg--
15:30:50.045 pjsua_call.c ..Unable to accept incoming call (no available account)
15:30:50.047 pjsua_core.c .RX 433 bytes Request msg ACK/cseq=11024 (rdata0x7fe8c4001c48) from UDP 172.20.1.8:5060:
ACK sip:056004@172.20.1.2:5060;ob SIP/2.0
Via: SIP/2.0/UDP 0.0.0.0:5060;rport;branch=z9hG4bKPj9599ad6d-2486-4483-8041-afac3bfdc05d
From: "MI 3" <sip:056003@172.20.1.8>;tag=e0d412a9-2d1d-45ef-be7f-04f0a8bdaf2e
To: <sip:056004@172.20.1.2;ob>;tag=z9hG4bKPj9599ad6d-2486-4483-8041-afac3bfdc05d
Call-ID: 1b79c22a-dc7c-4652-92ae-416f3eafcd82
CSeq: 11024 ACK
Max-Forwards: 70
User-Agent: FPBX-15.0.23(16.16.1)
Content-Length: 0
The agent is able to register and I can see both agents in the asterisk cli, but not sure why the other agent is not able to place the call to this one.
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