From 0b84b386b945778d374f915729fe2700a0f80b0a Mon Sep 17 00:00:00 2001 From: Olle Johansson Date: Thu, 1 Feb 2007 20:43:49 +0000 Subject: 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 --- channels/chan_sip.c | 9 ++++++++- configs/sip.conf.sample | 7 +++++++ 2 files changed, 15 insertions(+), 1 deletion(-) 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, "")); 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 -- cgit v1.2.3