diff options
| author | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-09-10 18:40:49 +0000 |
|---|---|---|
| committer | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-09-10 18:40:49 +0000 |
| commit | a70eed0cbe98b8dcd5f4bf8a0a30d05f9df0b8bd (patch) | |
| tree | 03df76b20020a7f4a8f73f7119d83fcc0a1d6c52 | |
| parent | dafdaca1651eb51a2179c88fc956d6e59fc60e0e (diff) | |
Merged revisions 216805 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
........
r216805 | oej | 2009-09-07 18:08:08 +0200 (MÃ¥n, 07 Sep 2009) | 2 lines
Since it's possible to have more than 999 calls, I'm changing the call counter roof to something higher.
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@217665 f38db490-d61c-443f-a65b-d21fe96a405b
| -rw-r--r-- | channels/chan_sip.c | 8 |
1 files changed, 4 insertions, 4 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 4769e0800..25d26008f 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -1085,10 +1085,10 @@ static int global_relaxdtmf; /*!< Relax DTMF */ static int global_rtptimeout; /*!< Time out call if no RTP */ static int global_rtpholdtimeout; /*!< Time out call if no RTP during hold */ static int global_rtpkeepalive; /*!< Send RTP keepalives */ -static int global_reg_timeout; +static int global_reg_timeout; static int global_regattempts_max; /*!< Registration attempts before giving up */ static int global_callcounter; /*!< Enable call counters for all devices. This is currently enabled by setting the peer - call-limit to 999. When we remove the call-limit from the code, we can make it + call-limit to INT_MAX. When we remove the call-limit from the code, we can make it with just a boolean flag in the device structure */ static unsigned int global_tos_sip; /*!< IP type of service for SIP packets */ static unsigned int global_tos_audio; /*!< IP type of service for audio RTP packets */ @@ -22975,7 +22975,7 @@ static void set_peer_defaults(struct sip_peer *peer) peer->autoframing = global_autoframing; peer->qualifyfreq = global_qualifyfreq; if (global_callcounter) - peer->call_limit=999; + peer->call_limit=INT_MAX; ast_string_field_set(peer, vmexten, default_vmexten); ast_string_field_set(peer, secret, ""); ast_string_field_set(peer, remotesecret, ""); @@ -23290,7 +23290,7 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, str } else if (!strcasecmp(v->name, "callbackextension")) { ast_copy_string(callback, v->value, sizeof(callback)); } else if (!strcasecmp(v->name, "callcounter")) { - peer->call_limit = ast_true(v->value) ? 999 : 0; + peer->call_limit = ast_true(v->value) ? INT_MAX : 0; } else if (!strcasecmp(v->name, "call-limit")) { peer->call_limit = atoi(v->value); if (peer->call_limit < 0) |