FreePBX/sipsettings - Swedish @ Weblate

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Asterisk NAT setting:<br /> yes = Always ignore info and assume NAT<br /> no = Use NAT mode only according to RFC3581 <br /> never = Never attempt NAT mode or RFC3581 <br /> route = Assume NAT, don't send rport

Translation Swedish

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	Source	Translation
13	Allow Transports Reload
14	Allow transports to be reloaded when the PBX is reloaded. Enabling this is not recommended, and may lead to issues.
15	Allowing Inbound Anonymous SIP calls means that you will allow any call coming in form an un-known IP source to be directed to the 'from-pstn' side of your dialplan. This is where inbound calls come in. Although FreePBX severely restricts access to the internal dialplan, allowing Anonymous SIP calls does introduced additional security risks. If you allow SIP URI dialing to your PBX or use services like ENUM, you will be required to set this to Yes for Inbound traffic to work. This is NOT an Asterisk sip.conf setting, it is used in the dialplan in conjuction with the Default Context. If that context is changed above to something custom this setting may be rendered useless as well as if 'Allow SIP Guests' is set to no.
16	An Error occurred trying fetch network configuration and external IP address
17	An unknown port conflict has been detected in PJSIP. Please check and validate your PJSIP Ports to ensure they're not overlapping
18	Asterisk NAT setting:<br /> yes = Always ignore info and assume NAT<br /> no = Use NAT mode only according to RFC3581 <br /> never = Never attempt NAT mode or RFC3581 <br /> route = Assume NAT, don't send rport	Asterisk NAT-inställningar:<br /> yes = Ignorera alltid info och förutsätt NAT<br /> no = Använd NAT-läge enligt RFC3581 <br /> never = Använd aldrig NAT-läge eller RFC3581 <br /> route = Förutsätt NAT, sänd inte rport
19	Asterisk SIP Settings	Asterisk SIP-inställningar
20	Asterisk is currently using %s for SIP Traffic.
21	Asterisk: bindaddr. The IP address to bind to and listen for calls on the Bind Port. If set to 0.0.0.0 Asterisk will listen on all addresses. It is recommended to leave this blank.	Asterisk: bindaddr. IP-adressen att binda till och lyssna efter samtal på Bindporten. Om detta sätts till 0.0.0.0 kommer Asterisk att lyssna på alla adresser. Det är rekommenderat att lämna detta fält tomt.
22	Asterisk: canreinvite. yes: standard reinvites; no: never; nonat: An additional option is to allow media path redirection (reinvite) but only when the peer where the media is being sent is known to not be behind a NAT (as the RTP core can determine it based on the apparent IP address the media arrives from; update: use UPDATE for media path redirection, instead of INVITE. (yes = update + nonat)	Asterisk: canreinvite. ja: standard reinvites; nej: aldrig; nonat: Ett extra val för att tillåta omstyrning av mediaströmmen (reinvite) men endast när peer där strömmen skickas till är känd att inte vara bakom NAT (eftersom RTP kan bestämma det baserat på den synbara IP-adressen strömmen kommer från; update: använd UPDATE för mediaomstyrning i stället för INVITE. (yes = update + nonat)
23	Asterisk: externrefresh. How often to lookup and refresh the External Host FQDN, in seconds.	Asterisk: externrefresh. Hur ofta uppslag och uppdatering ska ske för extern FQDN i sekunder.

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Glossary

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Source information

Source string location
chansip.page.php:174
Source string age
6 years ago
Translation file
i18n/sv_SE/LC_MESSAGES/sipsettings.po, string 18
String priority
Medium