This option specifies a preference for which music on hold class this channel allow=ilbc see doc/rtp-packetization for framing options allow=ulaw Allow codecs in order of preference Message-Account in the MWI notify message vmexten=voicemail dialplan extension to reach mailbox sets the when sending MWI to phones with this bug. Enable this option to not get error messages ![]() buggymwi=no Cisco SIP firmware doesn't support the MWI RFC checkmwi=10 Default time between mailbox checks for peers notifymimetype=text/plain Allow overriding of mime type in MWI NOTIFY t1min=100 Minimum roundtrip time for messages to monitored hosts defaultexpiry=120 Default length of incoming/outgoing registration minexpiry=60 Minimum length of registrations/subscriptions (default 60) maxexpiry=3600 Maximum allowed time of incoming registrations tos_video=af41 Sets TOS for RTP video packets. tos_audio=ef Sets TOS for RTP audio packets. See doc/README.tos for a description of these parameters. and multiline formatted headers for strict international character conversions in URIs pedantic=yes Enable checking of tags in headers, Use "sip show domains" to list local domains domain=mydomain.tld Set default domain for this host names to some other SIP users on the Internet ability to place SIP calls based on domain Srvlookup=yes Enable DNS SRV lookups on outbound calls bindport is the local UDP port that Asterisk will listen onīindaddr=0.0.0.0 IP address to bind to (0.0.0.0 binds to all) Set this to your host name or domain nameīindport=5060 UDP Port to bind to (SIP standard port is 5060) Realms MUST be globally unique according to RFC 3261 realm=mydomain.tld Realm for digest authentication allowtransfer=no Disable all transfers (unless enabled in peers or users) allowguest=no Allow or reject guest calls (default is yes)Īllowoverlap=no Disable overlap dialing support. Active SIP peers will not be reconfiguredĬontext=default Default context for incoming calls reload chan_sip.so Reload configuration file sip show registry Show status of hosts we register with sip show users Show all SIP users (including friends) sip show peers Show all SIP peers (including friends) Useful CLI commands to check peers/users: SIP/proxyhostname/user or where the proxyhostname is defined in a section below If you define a SIP proxy as a peer below, you may call (Don't forget to enable DNS SRV records if you want to use this) SIP/devicename where devicename is defined in a section below. ![]() Syntax for specifying a SIP device in nf is Modifs:fichiers par défaut apres installation Voici mes fichiers sip.conf et nf que j'ai sauvegarder avant de commencer les Serveur ubuntu hardy (avec Bind9 et squid) : 192.168.0.1 Sur les postes winxp j'ai installé x-lite et je crois que j'ai bien fait les configurations à ce niveau Je ne veut pas utiliser d'autre carte, je veux juste que par les cables réseaux, les postes puissent communiquer j'ai installé astérisk sur le serveur et je n'arrive pas à le configurer et vu aussi que la plupart des docs sur le sujet son en anglais alors que j'y comprend pas grande chose alors un coup de main s.v.pĮn fait je voudrais faire en sorte que les postes winxp puissent s'appeler avec x-lite donc je dois leur attribué des numéros: Je voudrais expérimenter asterisk dans un réseau comprenant 4 poste winxp et un serveur ubuntu hardy desktop.
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