Я могу запускать AsteriskNOW виртуально в моей локальной системе. Кроме того, я использую Anveo на двух разных жестких телефонах SIP. Anveo работает нормально, указывая на то, что сеть прекрасно обрабатывает SIP.
Как использовать бесплатную пробную версию SIP-станции Sangoma?
Потому что, когда я ввожу учетные данные в мерцании:
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$
просто пытаюсь настроить мерцание с пробной версией Sangoma.
Конечно, отлично работает с CSipSimple, используя планшет Android, подключенный к той же сети с Wi-Fi.