diff options
| -rw-r--r-- | channels/chan_sip.c | 2 | ||||
| -rw-r--r-- | configs/sip.conf.sample | 7 |
2 files changed, 3 insertions, 6 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index ec30f1602..93bc400e3 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -21147,7 +21147,7 @@ static int reload_config(enum channelreloadreason reason) if (ast_parse_arg(v->value, PARSE_INADDR, &stunaddr)) ast_log(LOG_WARNING, "Invalid STUN server address: %s\n", v->value); externexpire = time(NULL); - } else if (!strcasecmp(v->name, "bindaddr")) { + } else if (!strcasecmp(v->name, "bindaddr") || !strcasecmp(v->name, "udpbindaddr")) { if (ast_parse_arg(v->value, PARSE_INADDR, &bindaddr)) ast_log(LOG_WARNING, "Invalid address: %s\n", v->value); } else if (!strcasecmp(v->name, "localnet")) { diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index 64adbf2c1..54f497328 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -75,11 +75,8 @@ allowoverlap=no ; Disable overlap dialing support. (Default is yes) ; asterisk.conf, it defaults to that system name ; Realms MUST be globally unique according to RFC 3261 ; Set this to your host name or domain name -bindport=5060 ; UDP Port to bind to (SIP standard port for unencrypted UDP - ; and TCP sessions is 5060) - ; bindport is the local UDP port that Asterisk will listen on -bindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all) - ; You can specify port here too, like 123.123.123.123:5080 +udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all) + ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) ; ; Note that the TCP and TLS support for chan_sip is currently considered |