Twinkle-Konfiguration für Sangoma

Ich kann auf meinem lokalen System AsteriskNOW virtualisiert laufen . Außerdem verwende ich Anveo auf zwei verschiedenen SIP-Festtelefonen. Anveo arbeitet fein, was darauf hinweist, dass das Netzwerk SIP Fein abwickelt.

Wie nutze ich die kostenlose Testversion von Sangoma SIP Station?

Denn wenn ich die Anmeldeinformationen in Twinkle eingebe:

Failed to create a UDP socket (SIP) on port 5060 strerror_r is not available: 98

thufir@dur:~/.twinkle$  thufir@dur:~/.twinkle$ cat freepbx.cfg  # USER user_name=<name> user_domain=trunktrial1.freepbx.com user_display=thufir user_organization= auth_realm=trunktrial1.freepbx.com auth_name= auth_pass=<password> auth_aka_op=00000000000000000000000000000000 auth_aka_amf=0000  # SIP SERVER outbound_proxy= all_requests_to_proxy=no registrar= register_at_startup=yes registration_time=3600 reg_add_qvalue=no reg_qvalue=1  # RTP AUDIO codecs=speex-wb,speex-nb,g711a,g711u,gsm ptime=20 out_far_end_codec_pref=yes in_far_end_codec_pref=yes speex_nb_payload_type=97 speex_wb_payload_type=98 speex_uwb_payload_type=99 speex_bit_rate_type=cbr speex_dtx=no speex_penh=yes speex_quality=6 speex_complexity=3 speex_dsp_vad=yes speex_dsp_agc=yes speex_dsp_aec=no speex_dsp_nrd=yes speex_dsp_agc_level=20 ilbc_payload_type=96 ilbc_mode=30 g726_16_payload_type=102 g726_24_payload_type=103 g726_32_payload_type=104 g726_40_payload_type=105 g726_packing=rfc3551 dtmf_transport=auto dtmf_payload_type=101 dtmf_duration=100 dtmf_pause=40 dtmf_volume=10  # SIP PROTOCOL hold_variant=rfc3264 check_max_forwards=no allow_missing_contact_reg=yes registration_time_in_contact=yes compact_headers=no encode_multi_values_as_list=yes use_domain_in_contact=no allow_sdp_change=no allow_redirection=yes ask_user_to_redirect=yes max_redirections=5 ext_100rel=supported ext_replaces=yes referee_hold=no referrer_hold=yes allow_refer=yes ask_user_to_refer=yes auto_refresh_refer_sub=no attended_refer_to_aor=no allow_xfer_consult_inprog=no send_p_preferred_id=no  # Transport/NAT sip_transport=auto sip_transport_udp_threshold=1300 nat_public_ip= stun_server= persistent_tcp=yes enable_nat_keepalive=no  # TIMERS timer_noanswer=30 timer_nat_keepalive=30 timer_tcp_ping=30  # ADDRESS FORMAT display_useronly_phone=yes numerical_user_is_phone=no remove_special_phone_symbols=yes special_phone_symbols=-()/. use_tel_uri_for_phone=no  # RING TONES ringtone_file= ringback_file=  # SCRIPTS script_incoming_call= script_in_call_answered= script_in_call_failed= script_outgoing_call= script_out_call_answered= script_out_call_failed= script_local_release= script_remote_release=  # NUMBER CONVERSION  # SECURITY zrtp_enabled=no zrtp_goclear_warning=yes zrtp_sdp=yes zrtp_send_if_supported=no  # MWI mwi_sollicited=no mwi_user= mwi_server= mwi_via_proxy=no mwi_subscription_time=3600 mwi_vm_address=  # INSTANT MESSAGE im_max_sessions=10 im_send_iscomposing=yes  # PRESENCE pres_subscription_time=3600 pres_publication_time=3600 pres_publish_startup=yes thufir@dur:~/.twinkle$  

Versuchen Sie einfach, die Sangoma-Testversion zu konfigurieren.

Natürlich funktioniert CSipSimple perfekt, wenn Sie ein Android-Tablet verwenden, das mit dem gleichen Netzwerk über WLAN verbunden ist.