Fritzbox und VoIP-Einstellungen
Verfasst: Do Dez 04, 2014 9:39 pm
Hallo Leute,
ich habe einige Einstellungen ausprobiert und bekomme das VoIP nicht hin. Welche Daten müssen unbedingt wo geändert werden, wie zum Beispiel dnsport usw.
Hier ein Auszug der export.datei (Sicherung) aus der Fritte 6490 (hier funktioniert das Einstellen der Daten nur über die export.datei - da vom Anbieter beschnitten). Hat jemand eine normale nicht beschnittene Fritte und kann mal in seine export schauen, wie wo was geändert werden muss?
Vielen Dank
Gruß Mario
_________________________________________________
voipcfg {
dnsport = 7077;
rtpport_start = 7078;
sip_srcport = 5060;
use_dqos = yes;
ua1 {
enabled = no;
username = "37560";
authname = "";
passwd = "hier mein Passwort";
registrar = "stitz.stellwerksim.de";
ttl = 1h;
sipping_enabled = yes;
sipping_interval = 280s;
name = "37560";
providername = "";
ims_client = no;
with_displayname = no;
read_from_displayname = yes;
dtmfcfg = dtmfcfg_rtp_or_inband;
rtpevent_keep_packetrate = yes;
register_failwait = 0w;
register_failwaitmax = 30m;
stunserver = "";
stunserverport = 3478;
use_internat_calling_numb = no;
is_nat_aware = no;
localip = 0.0.0.0;
protocolprefer = protocolprefer_ipv4only;
ignore_received_header = no;
always_clir = no;
clirtype = clir_star67star;
reject_anonymous_call_with_433 = yes;
colptype = colp_none;
clipnstype = clipns_off;
vad_enabled = no;
only_one_dialog = no;
presence_supported = no;
mwi_supported = no;
mwi_inmemoria = no;
ccbs_supported = no;
reg_support = regsupport_auto;
packetization = packetization_fixed;
tx_packetsize_in_ms = 20;
xrtp_periodic = 0;
reject_refer = yes;
sip_instance = no;
no_register_fetch = yes;
do_not_register = no;
only_call_from_registrar = yes;
invite_without_register_allowed = no;
outboundproxy = "no";
outboundproxy_without_route_header = no;
factory_3pty_uri = "";
no_hold_speech = yes;
dditype = ddi_none;
ddireception = "";
webui_trunk_id = "";
alias_head_number = "";
cfxsignaling = cfx_standard;
backup_wanted = no;
use_session_timer = no;
use_rport = yes;
add_rtpmap_for_all_codecs = no;
answer_only_one_codec = no;
without_annexb_no = no;
srtp_supported = no;
crypto_avp_mode = crypto_mode_avp_only;
use_488_for_no_t38 = no;
g726_via_rfc3551 = no;
no_g726_32_offer_with_pt2 = no;
g726_fixed_ptime30 = no;
dtmf_inband_on_g711g722 = no;
enable_3xx = yes;
t38_reinvite_from_remote = no;
use_t38version0 = no;
rtcp_xr_media_attribute = yes;
ptime_a_attribute = yes;
tones_and_announcements_for_service = yes;
read_p_asserted_identity_header = no;
route_always_over_internet = no;
sipiface = sipiface_automatic;
altc_attribute_rfc6947 = no;
send_extended_sip_client_info = yes;
gui_readonly = yes;
convertstate = 0;
use_dqos = no;
snmp_instance = 0;
}
ich habe einige Einstellungen ausprobiert und bekomme das VoIP nicht hin. Welche Daten müssen unbedingt wo geändert werden, wie zum Beispiel dnsport usw.
Hier ein Auszug der export.datei (Sicherung) aus der Fritte 6490 (hier funktioniert das Einstellen der Daten nur über die export.datei - da vom Anbieter beschnitten). Hat jemand eine normale nicht beschnittene Fritte und kann mal in seine export schauen, wie wo was geändert werden muss?
Vielen Dank
Gruß Mario
_________________________________________________
voipcfg {
dnsport = 7077;
rtpport_start = 7078;
sip_srcport = 5060;
use_dqos = yes;
ua1 {
enabled = no;
username = "37560";
authname = "";
passwd = "hier mein Passwort";
registrar = "stitz.stellwerksim.de";
ttl = 1h;
sipping_enabled = yes;
sipping_interval = 280s;
name = "37560";
providername = "";
ims_client = no;
with_displayname = no;
read_from_displayname = yes;
dtmfcfg = dtmfcfg_rtp_or_inband;
rtpevent_keep_packetrate = yes;
register_failwait = 0w;
register_failwaitmax = 30m;
stunserver = "";
stunserverport = 3478;
use_internat_calling_numb = no;
is_nat_aware = no;
localip = 0.0.0.0;
protocolprefer = protocolprefer_ipv4only;
ignore_received_header = no;
always_clir = no;
clirtype = clir_star67star;
reject_anonymous_call_with_433 = yes;
colptype = colp_none;
clipnstype = clipns_off;
vad_enabled = no;
only_one_dialog = no;
presence_supported = no;
mwi_supported = no;
mwi_inmemoria = no;
ccbs_supported = no;
reg_support = regsupport_auto;
packetization = packetization_fixed;
tx_packetsize_in_ms = 20;
xrtp_periodic = 0;
reject_refer = yes;
sip_instance = no;
no_register_fetch = yes;
do_not_register = no;
only_call_from_registrar = yes;
invite_without_register_allowed = no;
outboundproxy = "no";
outboundproxy_without_route_header = no;
factory_3pty_uri = "";
no_hold_speech = yes;
dditype = ddi_none;
ddireception = "";
webui_trunk_id = "";
alias_head_number = "";
cfxsignaling = cfx_standard;
backup_wanted = no;
use_session_timer = no;
use_rport = yes;
add_rtpmap_for_all_codecs = no;
answer_only_one_codec = no;
without_annexb_no = no;
srtp_supported = no;
crypto_avp_mode = crypto_mode_avp_only;
use_488_for_no_t38 = no;
g726_via_rfc3551 = no;
no_g726_32_offer_with_pt2 = no;
g726_fixed_ptime30 = no;
dtmf_inband_on_g711g722 = no;
enable_3xx = yes;
t38_reinvite_from_remote = no;
use_t38version0 = no;
rtcp_xr_media_attribute = yes;
ptime_a_attribute = yes;
tones_and_announcements_for_service = yes;
read_p_asserted_identity_header = no;
route_always_over_internet = no;
sipiface = sipiface_automatic;
altc_attribute_rfc6947 = no;
send_extended_sip_client_info = yes;
gui_readonly = yes;
convertstate = 0;
use_dqos = no;
snmp_instance = 0;
}