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I'm able to run AsteriskNOW virtualized on my local system. Also, I'm using Anveo on two different SIP hard phones. Anveo works fine, indicating that the network handles SIP fine.

Now, how do I utilize the free trial from Sangoma SIP Station?

Because when I enter the credentials into twinkle:

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$ 

just trying to configure twinkle with Sangoma trial.

Of course, works perfectly from CSipSimple using an Android tablet connected to the same network with wi-fi.

1 Answer 1

2

Another program already bound to 5060

While strerror_r was not available there, ostensibly to print error #98, strerror(98) here returns Address in Use.

I imagine the issue is that some other program (e.g. AsteriskNOW) is already listening on port 5060, so twinkle would have to be configured to bind to a different port number.

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