客製~VOIP Settings 的參數

目的 1：要新增一個參數名為 call_limit。

VOIP Settings 是用來設定 SIP/ IAX 的設定參數，這裡的修改會針對設定檔 additional_a2billing_sip.conf 裡的內容。
如果因系統需要增加頁面沒有的 SIP/IAX 的參數，可以參考以下作法：

1. 資料庫的 Table cc_sip_buddies 增加一個欄位，並且名稱命名為 call_limit。

mysql> ALTER TABLE cc_sip_buddies ADD call_limit;

2. 編輯 /var/www/html/A2Billing_UI/Public/form_data/FG_var_friend.inc
注意看 call_limit 的部份

getpost_ifset(array('defaultuser', 'subscribemwi', 'vmexten', 'cid_number', 'callingpres', 'usereqphone', 'incominglimit', 'subscribecontext', 'musicclass', 'mohsuggest',
'allowtransfer', 'autoframing', 'maxcallbitrate', 'outboundproxy', 'regserver', 'rtpkeepalive', 'call_limit', 'dbsecret ', 'regcontext ', 'sourceaddress ', 'mohinterpret ', 'mohsuggest ',
'inkeys ', 'outkey ', ' sendani ', 'fullname ', ' maxauthreq ', 'encryption ', 'transfer ', 'jitterbuffer ', 'forcejitterbuffer ', 'codecpriority ', 'qualifysmoothing ',
'qualifyfreqok ', 'qualifyfreqnotok ', 'timezone' , 'adsi')); 

選一個類似欄位的寫法，複製然後將名稱改成你要的。

 $HD_Form -> AddEditElement (gettext("CALL-LIMIT"),
                                                           "call_limit",
                                                           '',
                                                           "INPUT",
                                                           "size=10 maxlength=10",
                                                           "",
                                                           "",
                                                           "" , "", "", "", "", "",     "", "" );

注意 IF 的上下段，上段是 SIP；下段是 IAX

if ($atmenu=='sip') {
        $field_list_edit .= ', username, md5secret, nat, dtmfmode, canreinvite, callgroup, fromuser, fromdomain, insecure, mailbox, mask, pickupgroup, restrictcid, rtptimeout, rtpholdtimeout, ' .
                        'musiconhold,  cancallforward, defaultuser, subscribemwi, vmexten, callingpres, usereqphone, incominglimit, subscribecontext, musicclass, mohsuggest, allowtransfer, ' .
                        'autoframing, maxcallbitrate, outboundproxy, regserver, rtpkeepalive, call_limit';
} else {
        $field_list_edit .= ', trunk, dbsecret , regcontext , sourceaddress , mohinterpret , mohsuggest , inkeys , outkey ,  sendani , fullname ,  maxauthreq , encryption , transfer , jitterbuffer , ' .
                        'forcejitterbuffer , codecpriority , qualifysmoothing , qualifyfreqok , qualifyfreqnotok , timezone , adsi, requirecalltoken, maxcallnumbers, maxcallnumbers_nonvalidated';
}

3. 編輯 /var/www/html/A2Billing_UI/lib/Class.Realtime.php
注意文字 call_limit 的部份

 $this -> FG_QUERY_ADITION_SIP = 'name, accountcode, regexten, amaflags, callgroup, callerid, canreinvite, context, DEFAULTip, dtmfmode, fromuser, fromdomain, host, insecure, language, ' .
                                'mailbox, md5secret, nat, deny, permit, mask, pickupgroup, port, qualify, restrictcid, rtptimeout, rtpholdtimeout, secret, type, username, disallow, allow, musiconhold, regseconds, ' .
                                'ipaddr, cancallforward, fullcontact, setvar, lastms, regserver, defaultuser, auth, subscribemwi, vmexten, cid_number, callingpres, usereqphone, incominglimit, subscribecontext, ' .
                                'musicclass, mohsuggest, allowtransfer, autoframing, maxcallbitrate, outboundproxy, rtpkeepalive, call_limit';

                $this -> FG_QUERY_ADITION_IAX = 'name, accountcode, regexten, amaflags, callerid, context, DEFAULTip, host, language, mask, port, qualify, secret, username, disallow, allow, regseconds, ' .
                                'ipaddr, trunk, dbsecret, regcontext, sourceaddress, mohinterpret, mohsuggest, inkeys, outkey, cid_number, sendani, fullname, auth, maxauthreq, encryption, transfer, jitterbuffer, ' .
                                'forcejitterbuffer, codecpriority, qualifysmoothing, qualifyfreqok, qualifyfreqnotok, timezone, adsi, setvar, type, deny, permit, requirecalltoken, maxcallnumbers, ' .
                                'maxcallnumbers_nonvalidated';

NOTE:

＊ 要加的參數不能包含有 dash，因為 MySQL 的欄位名稱不支援。

call-limit 另一種作法

例如 SIP 用戶的帳戶(username)為 1234567

編輯 /etc/asterisk/sip_custom_post.conf

[1234567](+)
call-limit=5 

記得 reload asterisk。