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-rwxr-xr-xchannels/chan_sip.c8
-rwxr-xr-xconfigs/sip.conf.sample6
2 files changed, 10 insertions, 4 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 2824aeae6..63a0d2026 100755
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -613,7 +613,7 @@ static int sip_reloading = 0;
#define REG_STATE_REJECTED 4
#define REG_STATE_TIMEOUT 5
#define REG_STATE_NOAUTH 6
-#define REG_STATE_GAVEUP 7
+#define REG_STATE_FAILED 7
/* sip_registry: Registrations with other SIP proxies */
@@ -4527,8 +4527,8 @@ static int transmit_notify_with_sipfrag(struct sip_pvt *p, int cseq)
static char *regstate2str(int regstate)
{
switch(regstate) {
- case REG_STATE_GAVEUP:
- return "Gave up.";
+ case REG_STATE_FAILED:
+ return "Failed";
case REG_STATE_UNREGISTERED:
return "Unregistered";
case REG_STATE_REGSENT:
@@ -4610,7 +4610,7 @@ static int sip_reg_timeout(void *data)
/* Ok, enough is enough. Don't try any more */
/* We could add an external notification here...
steal it from app_voicemail :-) */
- r->regstate=REG_STATE_GAVEUP;
+ r->regstate=REG_STATE_FAILED;
} else {
r->regstate=REG_STATE_UNREGISTERED;
r->timeout = -1;
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 5a988d954..c852e1ccb 100755
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -82,6 +82,8 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
;compactheaders = yes ; send compact sip headers.
+;sipdebug = yes ; Turn on SIP debugging by default, from
+ ; the moment the channel loads this configuration
;
; If regcontext is specified, Asterisk will dynamically
@@ -121,6 +123,10 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; (instead of type=friend) if you have calls in both directions
;registertimeout=20 ; retry registration calls every 20 seconds (default)
+;registerattempts=10 ; Number of registration attempts before we give up
+ ; 0 = continue forever, hammering the other server until it
+ ; accepts the registration
+ ; Default is 10 tries
;callevents=no ; generate manager events when sip ua performs events (e.g. hold)
;---------------------------------------------- NAT SUPPORT ------------------------