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: https://origsvn.digium.com/svn/asterisk/trunk@53082 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 5d7be1d3778523f7e26dc210cd62b9a1a21c234d..24c23ab7d325810aed2269adc863e03f0d9b1ec8 100644 (file)
--- 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 faf86e33dbd7119dd10de6408dc13fdbd5af1600..83073996d67dbd9a19dfa6498fce7c805707e3b4 100644 (file)
--- 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