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authoroej <oej@f38db490-d61c-443f-a65b-d21fe96a405b>2007-02-01 20:43:49 +0000
committeroej <oej@f38db490-d61c-443f-a65b-d21fe96a405b>2007-02-01 20:43:49 +0000
commitf7031572b3bc236dd96d919350b21327641c6b26 (patch)
tree2762573bee73b40cc35ca0fe6d88465578d3747b
parent4204d0b00845fcb87d64b7a0fbe093bb813345bd (diff)
Implementing "busy-limit".
If you set call limit and busy limit, chan_sip will indicate BUSY for a device that has reached the busy limit and allow calls up to the call limit, allowing for call transfers (that generate a new call). If you only set call limit, chan_sip will not indicate BUSY until that limit is filled. This affects SIP subscriptions, call queues and manager applications. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@53082 f38db490-d61c-443f-a65b-d21fe96a405b
-rw-r--r--channels/chan_sip.c9
-rw-r--r--configs/sip.conf.sample7
2 files changed, 15 insertions, 1 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 5d7be1d37..24c23ab7d 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1095,6 +1095,7 @@ struct sip_peer {
int inRinging; /*!< Number of calls ringing */
int onHold; /*!< Peer has someone on hold */
int call_limit; /*!< Limit of concurrent calls */
+ int busy_limit; /*!< Limit where we signal busy */
enum transfermodes allowtransfer; /*! SIP Refer restriction scheme */
char vmexten[AST_MAX_EXTENSION]; /*!< Dialplan extension for MWI notify message*/
char mailbox[AST_MAX_EXTENSION]; /*!< Mailbox setting for MWI checks */
@@ -10207,6 +10208,8 @@ static int _sip_show_peer(int type, int fd, struct mansession *s, const struct m
ast_cli(fd, " VM Extension : %s\n", peer->vmexten);
ast_cli(fd, " LastMsgsSent : %d/%d\n", (peer->lastmsgssent & 0x7fff0000) >> 16, peer->lastmsgssent & 0xffff);
ast_cli(fd, " Call limit : %d\n", peer->call_limit);
+ if (peer->busy_limit)
+ ast_cli(fd, " Busy limit : %d\n", peer->busy_limit);
ast_cli(fd, " Dynamic : %s\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_DYNAMIC)?"Yes":"No"));
ast_cli(fd, " Callerid : %s\n", ast_callerid_merge(cbuf, sizeof(cbuf), peer->cid_name, peer->cid_num, "<unspecified>"));
ast_cli(fd, " MaxCallBR : %d kbps\n", peer->maxcallbitrate);
@@ -10294,7 +10297,8 @@ static int _sip_show_peer(int type, int fd, struct mansession *s, const struct m
astman_append(s, "VoiceMailbox: %s\r\n", peer->mailbox);
astman_append(s, "TransferMode: %s\r\n", transfermode2str(peer->allowtransfer));
astman_append(s, "LastMsgsSent: %d\r\n", peer->lastmsgssent);
- astman_append(s, "Call limit: %d\r\n", peer->call_limit);
+ astman_append(s, "Call-limit: %d\r\n", peer->call_limit);
+ astman_append(s, "Busy-limit: %d\r\n", peer->busy_limit);
astman_append(s, "MaxCallBR: %d kbps\r\n", peer->maxcallbitrate);
astman_append(s, "Dynamic: %s\r\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_DYNAMIC)?"Y":"N"));
astman_append(s, "Callerid: %s\r\n", ast_callerid_merge(cbuf, sizeof(cbuf), peer->cid_name, peer->cid_num, ""));
@@ -15405,6 +15409,9 @@ static int sip_devicestate(void *data)
} else if (p->call_limit && (p->inUse == p->call_limit))
/* check call limit */
res = AST_DEVICE_BUSY;
+ else if (p->call_limit && p->busy_limit && p->inUse >= p->busy_limit)
+ /* We're forcing busy before we've reached the call limit */
+ res = AST_DEVICE_BUSY;
else if (p->call_limit && p->inUse)
/* Not busy, but we do have a call */
res = AST_DEVICE_INUSE;
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index faf86e33d..83073996d 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -195,6 +195,11 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; for a device. When the call limit is filled, we will indicate busy. Note that
; you need at least 2 in order to be able to do attended transfers.
;
+; If you set the busy-limit in addition to the call limit, we will indicate busy
+; when we have a number of calls that matches busy-limit, but still allow calls
+; up to the call-limit. This allows for transfers while still having blinking
+; lamps and queues understanding that a device is busy.
+;
; For queues, you will need this level of detail in status reporting, regardless
; if you use SIP subscriptions. Queues and manager use the same internal interface
; for reading status information.
@@ -491,6 +496,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; videosupport videosupport
; maxcallbitrate maxcallbitrate
; rfc2833compensate mailbox
+; busy-limit
; username
; template
; fromdomain
@@ -524,6 +530,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;host=box.provider.com
;usereqphone=yes ; This provider requires ";user=phone" on URI
;call-limit=5 ; permit only 5 simultaneous outgoing calls to this peer
+;busy-limit=2 ; Signal busy at 2 or more calls
;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
; Call-limits will not be enforced on real-time peers,
; since they are not stored in-memory