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freepbc 呼出�{�待旉��修改

Wed, 17 Apr 2013 03:17:00 GMT

modules\core\functions.inc.php
查找 macro-dialout-trunk
然后
$ext->add($context, $exten, '', new ext_dial('${OUT_${DIAL_TRUNK}}/${OUTNUM}', '300,${DIAL_TRUNK_OPTIONS}')); // Regular Trunk Dial

$ext->add($context, $exten, 'skipoutnum', new ext_dial('${pre_num:4}${the_num}${post_num}', '300,${DIAL_TRUNK_OPTIONS}'));

300 �Ҏ(gu��)��你自��p��|�的旉��

八叶�?/a> 2013-04-17 11:17 发表评论

proxmox ve 2.2 elastix2.3模板制作

Tue, 06 Nov 2012 08:59:00 GMT

安装好proxmox ve 2.2
ssh �?proxmox ve

apt-get -y update
apt-get -y makedev
apt-get -y install build-essential make pve-headers-`uname -r`
cd /usr/src/
wget http://downloads.digium.com/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-current.tar.gz
tar zxfv dahdi-linux-complete-current.tar.gz
cd dahdi-linux-complete-*
make all
make install
make config
mkdir /etc/asterisk
service dahdi start
dahdi_genconf
vi /etc/dahdi/modules 全部�?��h��?br />
sed -i 's|ipt_REJECT ipt_tos ipt_limit ipt_multiport iptable_filter iptable_mangle ipt_TCPMSS ipt_tcpmss ipt_ttl ipt_length|ipt_REJECT ipt_tos ipt_TOS ipt_LOG ip_conntrack ipt_limit ipt_multiport iptable_filter iptable_mangle ipt_TCPMSS ipt_tcpmss ipt_ttl ipt_length ipt_state iptable_nat ip_nat_ftp|' /etc/vz/vz.conf
/etc/init.d/vz restart

//��用ipv6
/boot/grub/grub.cfg �?linux /vmlinuz-2.6.32-16-pve root=/dev/mapper/pve-root ro quiet 替换�?linux /vmlinuz-2.6.32-16-pve root=/dev/mapper/pve-root ro ipv6.disable=1 quiet

在elastix2.3机器�?参�?http://wiki.openvz.org/Creating_a_CentOS_5.0_Template

First, create a file called /tmp/exclude and add the following lines to it:

.bash_history

lost+found

/dev/*

/mnt/*

/tmp/*

/proc/*

/sys/*

/usr/src/*

tar --numeric-owner -czvf /tmp/centos-5.7-x86_64-elasitx_orig-image.tar.gz -X /tmp/exclude /

where <ARCH> represents the system architecture (i386 or
x86_64) and <DISTRO> represents the distribution (default, minimal, etc.).

scp /tmp/centos-5.7-x86_64-elasitx_orig-image.tar.gz root@ip:/var/lib/vz/template/iso

回到proxmox ve 机器使用 web�?strong>centos-5.7-x86_64-elasitx_orig-image 为模板创建vt虚拟�?nbsp; 假设 vm id �?100
ssh �?proxmox ve
cd /var/lib/vz/private/100
Edit the etc/inittab file and comment out the lines that respawn /sbin/mingetty on tty1 through tty6. Just put a # at the beginning of the line.

�?etc/inittab 文�g最后面加入
1:2345:respawn:/sbin/agetty tty1 38400 linux

Remove the etc/mtab file and then create a symbolic link by typing ln -s /proc/mounts etc/mtab
Remove all of the lines from etc/fstab except for the line that mounts /dev/pts
Edit etc/rc.d/rc.sysinit and comment out the line that starts /sbin/start_udev by placing a # at the beginning of the line.

mknod dev/ptmx c 5 2
mkdir dev/pts

cd dev
/sbin/MAKEDEV ttyp ptyp
cd ../

mknod dev/null c 1 3

mknod -m 644 dev/random c 1 8

mknod dev/urandom c 1 9
mknod dev/tty9 c 4 9

Create the var/lock/rpm folder.

Edit etc/sysconfig/network and set NETWORKING_IPV6 to no.
Add the following lines to etc/modprobe.d/blacklist.conf:
blacklist ipv6

blacklist net-pf-10

Disable any physical NICs by modifying the etc/sysconfig/network-scripts/ifcfg-ethX files (where X is the interface number starting from 0) and setting ONBOOT to no.

Check etc/sysconfig/init to see if PROMPT=yes, and then change to no. Otherwise when startup init script rc will enter interactive mode and wait there forever

vzctl set 100 --devnodes dahdi/pseudo:rw --save
vzctl start 100
vzctl enter 100
chkconfig haldaemon off
chkconfig dahdi off
chkconfig wanrouter off

rm -rf /etc/init.d/dahdi
rm -rf /etc/init.d/wanrouter

chkconfig acpid off
chkconfig kudzu off

chkconfig xfs off
chkconfig rpcidmapd off
chkconfig rpcgssd off
chkconfig nfslock off
chkconfig mdmonitor off
chkconfig ip6tables off

exit
cd /var/lib/vz/private/100
tar -czvf /var/lib/vz/template/cache/centos-5-x86_64-elasitx.tar.gz ./

八叶�?/a> 2012-11-06 16:59 发表评论

�?a2billing

Thu, 11 Oct 2012 02:56:00 GMT

A2Billing 安裝��?http://www.osslab.com.tw/VoIP/IP_PBX/%e8%bb%9f%e9%ab%94%e5%bc%8f_IP_PBX/Asterisk/Addons/A2Billing_%e5%ae%89%e8%a3%9d%e7%af%87

A2Billing ��理��?

http://www.osslab.com.tw/VoIP/IP_PBX/%e8%bb%9f%e9%ab%94%e5%bc%8f_IP_PBX/Asterisk_-_%e5%85%8d%e8%b2%bb_IP_PBX_%e7%b6%b2%e8%b7%af%e9%9b%bb%e8%a9%b1%e4%ba%a4%e6%8f%9b%e5%b9%b3%e5%8f%b0/Addons/A2Billing_%e7%ae%a1%e7%90%86%e7%af%87

1. 前言
2. 初始化設�?/a>
3. 應用一�Q�國際電話卡模式
4. 應用二：(x��)DID 轉接業務
5. 應用三：(x��)預付制、月付制會員模式
6. 應用四：(x��)整合 FreePBX 的應�?/a>
7. 應用五：(x��)Caller ID 認證模式
8. F.A.Q

前言

A2Billing 是目前社��開發最熱門的計�ȝ��i�套�Ӟ��因為是開攑֎�始碼授權所以可以合法免��M��用�?/p>

本篇主要在是教導如何使用這套�Ӟ��若還沒完成安裝的朋友�Q�請先前往��p�� A2Billing 安裝��?/a>�?/p>

初始化設�?/h4>

�pȝ��(d��ng)在完成安裝後�Q�第一個所要做的設定有�Q?/p>

// FreePBX UI > Trunks

新增 Outbound Trunk�Q�本��以 Voxalot ��Z��?/p>

Outgoing Settings
Trunk Name = voxalot

host=us.voxalot.com
username=
fromuser=
secret=
fromdomain=voxalot.com
nat=yes
insecure=port,invite
qualify=yes
canreinvite=yes
dtmfmode=auto
disallow=all
allow=ulaw&alaw
type=peer
context=from-trunk

新增 Inbound Trunk�Q�這個主要在展示國際電話卡及(qi��ng) DID 轉接的應用，本篇�?iptel ��Z��?/p>

Outgoing Settings
Trunk Name: iptel

username=
type=friend
secret=
qualify=yes
insecure=port,invite
host=iptel.org
fromuser=osslab
fromdomain=iptel.org
context=a2billing

注意�Q�context 必須�?a2billing

// A2Billing Admin UI > Trunk > Create Provider

provider name = VOXALOT_PROVIDER
description = Voxalot Provider

// A2Billing Admin UI > Trunk > Add Trunk

voip-provider = VOXALOT_PROVIDER
label = VOXALOT_TRUNK
add prefix = �I�白
provider tech = SIP
provider ip = voxalot
status = Active

TIPs:

provider ip 必須�?FreePBX �?trunk name 相同

// A2Billing Admin UI > Ratecard > Create call plan

name = VOXALOT_CALLPLAN
remove inter prefix = YES

TIPs:

remove inter prefix = YES 若撥號有包含 00 �?011 開頭的國際冠��|��在套�?ratecard 的規則前會被去除�?/p>

// A2Billing Admin UI > Ratecard > Create new ratecard

tariffname = VOXALOT_RATECARD
trunk = VOXALOT_TRUNK
description = through voxalot trunk

// A2Billing Admin UI > Ratecard > Add Rate

ratecard = VOXALOT_RATECARD
dialprefix = 1800
destination = US-Tollfree
buying rate = 1
buyrate min duration = 6
buyrate billing block = 6
selling rate = 1.5
sellrate min duration = 60
sellrate billing block = 60
trunk = VOXALOT_TRUNK

TIPs:

�Q?dialprefix 若撥號規則符合，會以此費率計��?br />�Q?destination 用來敘述此費率的區�?/p>

//新增 rate 後要再回�?call plan ��?rate card 加入�?call plan
A2Billing Admin UI > Ratecard > List Call Plan > Edit: VOXALOT_CALLPLAN

ratecard = 選擇 VOXALOT_RATECARD, Add Ratecard

應用一�Q�國際電話卡模式

客戶操作��程�Q?br />① 使用手機或一般電話機撥打�pȝ��(d��ng)號碼(Access Number)

② 語音提示�Q��入電話卡號＋�Q?/p>

③ 語音提示��額�Q��入目的地電話號碼�Q?0�Q�國��|��區��|��電話號碼�Q�＃�Q?/p>

// A2Billing Admin UI > Customers > Create Customers

card number = <隨機產生>
card alias = <隨機產生>�Q�web ��d��希��
webui password = <隨機產生>�Q�web ��d��密碼
balance = 50.0�Q�不可為�Ӟ��且必須以 USD �a�算
call plan = VOXALOT_CALLPLAN
activated = YES
simultaneous access = INDIVIDUAL ACCESS
card type = PREPAID CARD
country = TAIWAN
sip account = NO
iax account = NO

實際操作�Q?/p>

因為本篇實做是以 iptel 的號��g��為系�i�p��|��所以用其他 IPTel 希��撥入�pȝ��(d��ng)號碼�Q�系�i�會提示語音�Q�請輸入 card number�Q�將剛剛新增的卡號��入，完成後按�Q�，若正��，�pȝ��(d��ng)會提�C�餘��及(qi��ng)可用通話時間�Q�並提示�Q�請輸入目的地電��p��|��完整電��p��?011 + 國碼 + 區��?+ 電話號碼輸入�Q�完成後按＃�?/p>

TIPs:

�Q?新增電話�?Customer)�Q�有一些主要的�a�定�Q�例�?card number, card alias, password�Q�這些是由�pȝ��(d��ng)隨機產生�Q�且沒有��M��規則�Q�hacker 無法事先�a�算出這些資訊�?br />�Q?附加�?card number �?另一�i�代�?alias�Q�這個號��或 email 地址都可以用來登入用戶的�E�頁�Q�密��就�?webui password�?
�Q?若用戶撥號時有加 00�Q�請注意 Call Plan �?REMOVE INTER PREFIX ��a�為 YES�Q�以避免 outbound 路由錯誤�?/p>

應用二：(x��)DID 轉接業務

A2Billing 提供 DID 接駁至經�?outbound trunk 的外部號��{��SIP URI�?/p>

// A2Billing Admin UI > DID > Add DID Group

name = DID TWN

// A2Billing Admin UI > DID > Add DID

DID = 99474
billing = only dialout rate
DID group = DID TWN
country = TAIWAN
activated = YES
monthly rate = 0

// A2Billing Admin UI > Customers > Create Customers

balance = 50.0
call plan = VOXALOT_CALLPLAN
didgroup = DID TWN
activated = YES
card type = PREPAID CARD
country = TAIWAN
sip account = NO
iax account = NO

�a�定轉接的目的號��|��可以透過 Admin UI �?Customer UI 來作
客戶自行��D�� DID 號碼
// A2Billing Customer UI > DID

select country = TAIWAN
select virtual phone number = 99474
voip call = NO
destination = 8864123456

TIPs:

�Q?select country 這裡內容會與 DID �?country 相同
�Q?若沒有出�?DID number 可選擇，請檢查所��d��?customer 希��?DIDGROUP 是否正確
�Q?voip call 若是 NO�Q��入外部的 PSTN/SIP 號碼�Q�若�?YES�Q��?SIP URI�?/p>

��理員分�?DID 號碼�i�指定的客戶
// A2Billing Admin UI > INBOUND DID > Destination

destination = 8864123456
customer ID = 選擇適當�?customer
DID = 選擇適當�?DID
activated = yes
validated = 皆可
voip_call = no

TIPs:

�Q?voip call 若是 NO�Q��入外部的 PSTN/SIP 號碼�Q�若�?YES�Q��?SIP URI�?/p>
�Q?若需 DID 接入內部分機號碼(不經�?A2B �?trunk)�Q�可以這樣�a?br />-- destination = Local/101@from-internal (適用 FreePBX 的分��?101)
-- destination = SIP/123456 (適用 A2Billing card number 123456)
-- voip_call = yes

* 如果 destination number 要走 A2B �?trunk�Q�不��該 trunk �?SIP �?Zaptel�Q�voip_call 必須�?no�?/p>

盔R��文章連結�Q?/p>

Elastix Forum :: Inbound DID Billing for A2Billing

應用三：(x��)預付制、月付制會員模式

用戶操作��程�Q?br />① 用戶使用 X-Lite �?ATA �a�備透過�E��\�a�d��為分��?/p>

② 直接撥目的地電話號碼�Q�例�?886+XXXXXXXX�Q?86 是國��{�?br />過程中不會有��M��外的提�C�音�Q�就像一般的 SIP 分機操作相同�?/p>

// A2Billing Admin UI > Customers > Create Customers

TIPs:

�Q?這裡�?應用一)模式�a�定相同�Q�只差在 sip/iax account �?YES�?br />�Q?新增 customer 後，�?reload Asterisk 後，SIP 用戶端才能註冊��用�?/p>

// 取消 "輸入目的地號��? �?qi��ng)其他提�C�音
// A2Billing Admin UI > System Settings > Global List
如果這裡的所有項目沒�?GROUP �?agi-conf2 �?可善�?GROUP 搜尋功能)�Q�請先前往 Add agi-conf > CREATE AGI-CONF2�?/p>

回到 Global List 扑ֈ� GROUP �?agi-conf2 �?KEY/VALUE 按以下所�q�C��改：(x��)

use_dnid = yes

這個SIP希��除了(ji��n)要撥外部的�\由外�Q�如果還要撥內部的其�?SIP 分機號，需要繼�U�下面的修改�Q?/p>

sip_iax_friends = yes
sip_iax_pstn_direct_call_prefix = 555
sip_iax_pstn_direct_call = yes

※是否要提�C�餘��的語音

say_balance_after_auth = no
say_timetocall = no

如果電話不通，不要提示輸入目的地號��的語音

number_try = 1

TIPs:

�Q?參數說明�Q?/p>
- Use DNID : YES 表示使用 DNID�Q�並且不會提�C?輸入目的地電��p��的語音
- SIP Call = yes, SIP Call Prefix, Direct Call: 這三個參數是用來�?SIP �a�d��希��後，也能撥其他分��號��|��撥法�?555+SIP Extension
- Say Balance After Auth: 認證後是否提�C�餘��?br />- Say Duration: 是否提示剩餘通話時間

- 另一�E�比較快速的�a�定是：(x��)play_audio = no, use_dnid=yes, number_try=1�?/p>

//為此模式新增 dialplan
//�R�輯 /etc/asterisk/extensions_a2billing.conf�Q�在底下加入�Q?/p>

[custom-a2billing-sipclient]
exten => _X.,1,Answer
exten => _X.,n,Wait(1)
exten => _X.,n,deadAGI(a2billing.php|2)
exten => _X.,n,Hangup

// �?A2Billing 套用新增�?dialplan
// A2Billing Admin UI > Customer > VoIP Settings
扑ֈ� SIP 分機項目�Q�按�R�輯

Context = custom-a2billing-sipclient

// 最後，埯��指��o(h��)套用所有的修改

# asterisk -rx "reload"

TIPs:

由於版本 1.7.x �?bug 關係�Q�在 UI 做完操作後，並不會同步修�?asterisk 的設定檔�Q�所以請依照方式作手動修改：(x��)
�R�輯 /etc/asterisk/additional_a2billing_sip.conf �Q�修�?context 的內宏V�?br />要避免每�ơ都要作手動修改�Q�可以參�?F.A.Q 的方法�?br />

應用四：(x��)整合 FreePBX 的應�?/h4>

用途：(x��)FreePBX 的分��用�Ӟ��?Outbound Calling 時可以透過 A2Billing 來計費，但其他原有PBX的功能都不會有媄(ji��ng)�ѝ�?/p>

// �R�輯 /etc/asterisk/extensions_custom.conf

[macro-dialout-trunk-predial-hook]
exten => s,1,GotoIf($["${OUT_${DIAL_TRUNK}:4:4}" = "A2B/"]?custom-freepbx-a2billing,${OUTNUM},1:2)
exten => s,2,MacroExit

[custom-freepbx-a2billing]
exten => _X.,1,DeadAGI(a2billing.php,${OUT_${DIAL_TRUNK}:8})
exten => _X.,n,Hangup()

// FreePBX UI > Trunks > Add Custom Trunk

Custom Dial String = A2B/2

Tips:

2 代表�?agi-conf 2

// FreePBX UI > Outbound Routes
自行�a�定需要的參數�Q�並��向剛剛�?Trunk�?/p>

// FreePBX UI > Extensions
在需要計�ȝ��分機�a�定裡，參數 accountcode 填入 A2Billing �?Card Number�?/p>

Tips:

分機在外撥的路由規則�Q�除�?outbound route 外，還要考慮 A2Billing 的，也就�?RATES 的相關設定�?/p>

延��p��

Using A2Billing to account for extension calls in Elastix (A2Billing 1.3.x)

應用五：(x��)Caller ID 認證模式

用途：(x��)�a�定電話卡時除了(ji��n)使用輸入卡號的認證方式以外，還能以來電號��?Caller ID) 辨識?sh��)��認證�?/p>

啟用 Caller ID 後，用戶不需要先輸入卡號�Q�就可以直接撥目的地號碼�?/li>
一個卡號可以設定一�i�或多組的來電號��{�?/li>
使用 Admin UI 來管理來電號��{�?/li>
一旦啟用後�Q�當用戶的來電號��g��非系�i�所允許時，�pȝ��(d��ng)會提�C�卡號的認證的方式�?/li>

// 啟用方式
A2Billing Admin UI > Syetem Settings >

�?agi-conf 的方式來�a�，可以很容易啟�?關閉這功能，需要用到的參數(Key)如下�Q?/p>

cid_enable = yes �Q�yes 啟用�Q�no 關閉�Q�預�a�是關閉�?/li>
cid_askpincode_ifnot_callerid = yes ; 預設 yes�Q�若 CID 認證失敗�Q�系�i�會提示輸入 Card Number�?/li>
cid_auto_assign_card_to_cid = yes ; 預設 yes�Q�用戶撥入系�i�後�Q�如�?CID 認證失敗�Q�系�i�會提示輸入 Card Number�Q�一旦用戶��入資�a�正��，�pȝ��(d��ng)會自動新�?CID�Q�以致於該用戶下�ơ再撥入時，��可以通過 CID 認證�Q�且不會再提�C��?Card Number�?/li>
cid_auto_create_card = no

// 新增 Caller ID

A2Billing Admin UI > Customers > Caller-ID

CallerID = <用戶的來電號��?gt;
Activated = yes
ID Card = <選擇卡號>

F.A.Q

Q:如何大量刪除 Customers?

Ans: A2Billing UI > Customers > Add::Search > Search Customers

�a�定搜尋條�g > �?Search > 再按旁邊�?Delete All

注意�Q�這個方法無法將 VoIP Setting 一併刪除�?/p>

Q:如何大量刪除 VoIP Settings?

Ans: 目前沒有合適的作法�?/p>

Q:如何變更�pȝ��(d��ng)預設�q�別 USD

Ans: 要改兩個地方，修改 base_currency 的參數及(qi��ng)更新 currency list�?/p>

A2Billing UI > System Settings > Global List
搜尋 GROUP = global, Key = base_currency
base_curreny = twd
即��修改�?ji��n)幣別，�pȝ��(d��ng)預設�?currency list 仍是以美�?1:1 �a�算�Q�所以必須修攚w��個匯率對照表�?TWD 1:1�?br />A2Billing UI > BILLING > Currency List > CLICK HERE TO UPDATE NOW

Q:匯入 ratecard 時總是出��N��?ERROR: file type is not allowed: application/force-download

Ans: ��副檔名 csv �Ҏ(gu��)�� txt�?/p>

盔R��連結

Asterisk: optimizing a2billing mysql database with more than 1,000,000 rates in cc_ratecard
一些有用的教學(英文)

發現 Bug

版本	問題描述
1.7.0/1	修改 VoIP Settings �?內容�Q�不會同步更�?Asterisk �a�定
	Ans:這是因為 A2Billing 預設是啟動了(ji��n) Realtime Asterisk 模式(Asterisk 以資料��n型式儲存�a�定)�Q�若只是檢查 .conf 來判�?Asterisk 是否更新是不準確的，然而�?bug 卻造成 A2Billing 無法�?relatime 模式更新 Asterisk。暫時的解決�Ҏ(gu��)��是�� A2Billing 關閉 realtime asterisk 模式�Q�恢復成 .conf 方式來更�?asterisk�Q�步驟如下：(x��) A2Billing UI > System Settings > Global List 搜尋 GROUP = global use_realtime = no (預設�?yes) Notes: 畉��閉了(ji��n) Realtime 後，爑־�若有新增/�R�輯 VoIP Settings 時，�?CONFIRM DATA 之後�Q�需要在作以下步驟完�?Asterisk 的更斎ͼ�(x��) 點選 GENERATE ADDITIONAL_A2BILLING_SIP.CONF click here to reload your asterisk server
1.7.0	新增 CUSTOMER 時，�?balance �Ƅ位�E�持預設�?0�Q�仍可以存檔
	Ans: 這項不確定是否為 bug�Q�但�?1.3.x �I�定版，存檔前系�i�會檢查 balance 不可�?0�?
1.7.0	若徏立電話卡是有 SIP �a�d��希��的，刪除這個電話卡後，SIP �i�端仍可以註冊，但已無法正常撥出�?/td>
1.7.0	�?Customer UI �?DID 功能�Q�操�?Release DID�Q�按�?Release 後，雖然 DID �?release�Q�但畫面會成�I�白頁�?/td>
1.7.0	�?Customer UI �?DID 功能�Q�從項目列表中刪�?destination number 時，所有欄位的值會出現�I�白�Q�再按下 Delete 後，雖然資料仍會刪除�Q�但需要再做一��?Release DID�?/td>

八叶�?/a> 2012-10-11 10:56 发表评论

asterisk 调试讄��

Fri, 13 Jul 2012 23:39:00 GMT

Logging

Sometimes, when debugging an issue, it's useful to see and log extra information and at

other times, you want logging to be minimal. Asterisk provides a number of ways of

logging information, to files or to a syslog server. The file /etc/asterisk/logger.conf

contains the configuration elements for logging. Asterisk has different types of message

that can be logged these are:

Verbose

General 'chatter' about what is

happening on the system.

Verbosity levels greater than 3

display dialplan commands as

they are executed. This

generates lots of log information

Debug

Debug messages, normally only

used by programmers to extract

extended information

Notice

Non urgent alert messages

Warning

Warning alert messages,

something happened that might

be bad. Some tell you how bad

the warning is

Error

Error messages, something bad

happened – These should be rare.

In logger.conf you will see the [logfiles] section, this is where you define the filename

and content of log files. Take a look at this example entry

[logfiles]

debug => debug

This tells Asterisk to log debug messages (the right side of the =>) to a file called debug

(the left side of the =>) located in /var/log/asterisk/. This directory can be changed in

/etc/asterisk/asterisk.conf by modifying the line astlogdir => /var/log/asterisk to point

to the desired directory. You can log multiple information types to the same file or you

can spread the information over a number of files. For example

[logfiles]

debug => debug

messages=>warning, error

Will log debug messages to a file called debug, and will also log warning and error

messages to a file called messages. There is a special “file” called console which when

used will cause the message types specified to be logged to the Asterisk console, for

example:

[logfiles]

console => debug, warning, error, notice, verbose

Would log everything to the console (not to any files). The above is not recommended

since the amount of information that would be generated could be far to much to be of

any real use. If you change logger.conf you need to perform a reload or do a logger

rotate (see next section). You can also log messages to a syslog sever (remote logging

server), useful if you have either a lot of machines or want centralized logging. To do

this use the file syslog, with the suffix you will use in /etc/syslog.conf for example, in

logger.conf

[logfiles]

syslog.local0 => debug, warning, error, notice, verbose

And /etc/syslog.conf

local0.* @192.168.1.22

This would send the Asterisk logging information to the syslog server at 192.168.1.22.

Setting up a syslog server is beyond the scope of this document and is left as an exercise

for the reader.
Rotate logs

It is advisable to rotate your logs frequently, depending on the amount of logging you

have turned on and the about of data that is actually logged. Files larger than 2Gb can

cause some nasty effects resulting in Asterisk crashes. You can rotate logs by using the

command

logger rotate

This will rename the old log and start a new one. It will also reload logger.conf and

adopt any changes you have made to it.

Verbosity

You can change the verbosity (how much information we get) of the output on processes attached to the Asterisk console by setting the level of verbosity. To do this we use the set verbose command, for example:

set verbose 999

Sets the verbosity level to 999, Asterisk will tell you that the level of verbosity changed

asterisk*CLI> set verbose 999

Verbosity is at least 999

asterisk*CLI>

You should see every message when it is set to this level, whereas setting it to 1 will

show very little information.

Setting the verbosity level changes the level on every attached

process (connected via asterisk -r) not just the one you issue the

command from.

Getting to Grips with sip.conf

sip.conf is not difficult to understand, however there is a fundamental problem with SIP making it awkward to use. The problem is not so much with SIP itself, it's more to do with how we protect our networks. We all know that there are some nasty little people out there who are quite happy to steal your resources, make free calls using someone else's money etc. As a consequence of this we tend to put a firewall between them and us. This is where the problems start for SIP. In a network environment that requires no firewall, for example and Internal network, there will be no issues, but in a more complex network, perhaps using NAT (Network address translation) there are all sorts of hurdles to overcome.

You have a number of choices,

sip set debug on 讄��昄��更多的sip信息
sip set debug off关闭昄��更多的sip信息
SIP Channels are only shown if registered
SIP SHOW INUSE will list all SIP extensions defined in SIP.CONF
SIP SHOW CHANNELS will list all SIP extensions registered at that time
sip set debug - Enable SIP debugging
sip set debug ip - Enable SIP debugging on IP
sip set debug off - Disable SIP debugging
sip set debug peer - Enable SIP debugging on Peername

stop gracefully 优雅地停止asterisk
stop now 立即停止�q�行asterisk
core show codecs 昄��所有支持的�~�解码器

set verbose 10
set debug 10

core set debug channel - Enable/disable debugging on a channel
core set debug - Set level of debug chattiness
core set debug off - Turns off debug chattiness
core set verbose - Set level of verboseness

core show calls {uptime}: Display information on calls

logger set level {DEBUG|NOTICE}: Enables/Disables a specific logging level for this console
logger set level debug off

八叶�?/a> 2012-07-14 07:39 发表评论

Wed, 27 Jun 2012 07:46:00 GMT

参�?br />http://brunoplum.iteye.com/blog/593263

yum install libxml2-devel
yum install ncurses-devel
yum install openssl-devel
yum install zlib-devel
yum install mysql-devel

wget http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/certified-asterisk-1.8.11-cert2.tar.gz
tar zxvf certified-asterisk-1.8.11-cert2.tar.gz
cd certified-asterisk-1.8.11-cert2
./configure
make menuselect (�H�口太小可能弹不出选择界面从新选择 make distclean ./configure)
add-ons 选择 res_config_mysql app_mysql cdr_mysql
applications 增加选择 app_voicemail
channel drivers 增加选择 chan-dahdi

vi apps/app_queue.c

try_calling((struct queue_ent *qe, 下面
char k_exten[15] = "";;
int k9i = 0;
int k9j = 0;
int k9_flag = 0;

bridge = ast_bridge_call(qe->chan,peer, &bridge_config); 前面
for(;k9i < 15;k9i++){
    if(member->interface[k9i]=='/'){
        k9_flag=1;
        continue;
    }
    if(member->interface[k9i]=='@'){
        k9_flag=0;
        break;
    }
    if(k9_flag){
        k_exten[k9j++]=member->interface[k9i];
    }
}
play_file(qe->chan, "number-report-begin");
ast_say_digit_str(qe->chan, k_exten, AST_DIGIT_ANY, qe->chan->language);
play_file(qe->chan, "number-report-end");
play_file(peer, "beep");

make
make install

vi /etc/asterisk/queues.conf
[general] 下面加入
setinterfacevar=yes

/Files/byc/wav.zip

八叶�?/a> 2012-06-27 15:46 发表评论

星网锐捷 IDA 讑֤� 查看IP

Wed, 23 May 2012 08:43:00 GMT

使用�?sh��)话��接第一个FXS�?br />拨打 #190# 听到�Ҏ(gu��)��提示��x(ch��ng)��?br />再拨�?#120# ��可以听到语��x(ch��ng)��报IP �?br />默认IP 192.168.88.16

八叶�?/a> 2012-05-23 16:43 发表评论

elastix2.3 dashboard 无交换空间时不显�C?bug 补丁

Tue, 22 May 2012 14:06:00 GMT

paloSantoDataApplets.class.php 416�?
//SWAP USAGE
if($arrSysInfo['SwapTotal']==0){
  $swap_usage_val = number_format(0.0,1);
}
else{
  $swap_usage_val = number_format(100.0 * ($arrSysInfo['SwapTotal'] - $arrSysInfo['SwapFree'])/$arrSysInfo['SwapTotal'], 1);
}
paloSantoSysInfo.class.php 157�?br />        if($this->arrSysInfo['SwapTotal']==0)
        {
                $value = 0;
        }
        else
        {
                $value = ($this->arrSysInfo['SwapTotal'] - $this->arrSysInfo['SwapFree'])/$this->arrSysInfo['SwapTotal'];
        }

八叶�?/a> 2012-05-22 22:06 发表评论

elastix2.3 中文语言�?voip88发布的elastix2.3中文版中抽取出来�?

Tue, 22 May 2012 12:36:00 GMT

使用�Ҏ(gu��)�� 解压后直接覆�?br />下蝲链接�Q?a href="/Files/byc/cn_elastix.2.3.zip">cn_elastix.2.3.zip
原始版本�Q?a >http://bbs.voip88.com/thread-27724-1-1.html

八叶�?/a> 2012-05-22 20:36 发表评论

elastix 2.3 升��到php5.2 CDR Report 出错 patch

Tue, 15 May 2012 12:44:00 GMT

html/libs/paloSantoDB.class.php

163 �?�Q�函�?fetchTable 内）(j��)
$r = $result->execute($param);
改�ؓ(f��)
if(isset($param))
{
foreach($param as $key => $value)
{

           if(is_int($value))
                     $paramtype = PDO::PARAM_INT;
                 elseif(is_bool($value))
                     $paramtype = PDO::PARAM_BOOL;
                 elseif(is_null($value))
                     $param = PDO::PARAM_NULL;
                 elseif(is_string($value))
                     $paramtype = PDO::PARAM_STR;
                 else
                     $paramtype = FALSE;

                if($paramtype )
                     $result->bindValue($key+1,$value,$paramtype);
         }
}
$r = $result->execute();

八叶�?/a> 2012-05-15 20:44 发表评论

elastix (Monitoring -> listen )ie 听录�?补丁

Wed, 25 Apr 2012 09:18:00 GMT

/var/www/html/modules/monitoring/index.php 423�?br />

CLASSID="CLSID:22D6f312-B0F6-11D0-94AB-0080C74C7E95"
TYPE="application/x-oleobject"
CODEBASE="

NAME="MediaPlayer" WIDTH=300 HEIGHT=100
autostart="true" showcontrols="true" showstatusbar="true">

八叶�?/a> 2012-04-25 17:18 发表评论

freepbx-mysql密码修改

Fri, 24 Feb 2012 09:21:00 GMT

http://www.freepbx.org/support/documentation/faq/changing-the-mysql-password

amportal.conf, cdr_mysql.conf, res_mysql.conf freepbx.conf

Once the password works, you need to update three files:

/etc/amportal.conf:

AMPDBUSER=asteriskuser

AMPDBPASS=mypass

/etc/asterisk/cdr_mysql.conf:

password=mypass

user=asteriskuser

/etc/asterisk/res_mysql.conf:

dbuser = asteriskuser

dbpass = mypass

./etc/freepbx.conf
$amp_conf['AMPDBPASS'] = 'mypass';

八叶�?/a> 2012-02-24 17:21 发表评论

asterisk-cpp

Fri, 06 Jan 2012 03:22:00 GMT

��目地址 https://github.com/augcampos/asterisk-cpp
asterisk-cpp 功能部分�q�可以，但bug 实在太多�Q��ƈ发现�?设计�l�构不合�?产生的bug,不徏议��用�?br />

八叶�?/a> 2012-01-06 11:22 发表评论

asterisk-dotnet 1.6.3.1 Asterisk 1.6.2.21 一直重复连�? BUG

Fri, 09 Dec 2011 08:47:00 GMT

asterisk-dotnet 1.6.3.1 Fix Ping actionid with 1.6.0.10 version
asterisk 1.6 现在的版本是 Asterisk 1.6.2.21
问题 asterisk 1.6.0.10 �?Ping 命��o(h��)�q�回时没有actionId�Q�但Aserisk 1.6.2.21Ping 命��o(h��)�q�回却有actionId�?a >

详细�?�Q�https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=13993
补丁�Q?br />ManagerConnection.cs 1982 行改�?br />

            if (!string.IsNullOrEmpty(actionId))
            {
                int hash = Helper.GetInternalActionId(actionId).GetHashCode();
                responseActionId = Helper.StripInternalActionId(actionId);
                responseHandler = GetRemoveResponseHandler(hash);

                if (response != null)
                    response.ActionId = responseActionId;
                if (responseHandler != null)
                {

                }
                else if (response == null && buffer.ContainsKey("ping") && buffer["ping"] == "Pong")
                {
                    response = Helper.BuildResponse(buffer);
                    foreach (ResponseHandler pingHandler in pingHandlers.Values)
                        pingHandler.HandleResponse(response);
                    pingHandlers.Clear();
                }
            }
            else if (response == null && buffer.ContainsKey("ping") && buffer["ping"] == "Pong")
            {
                response = Helper.BuildResponse(buffer);
                foreach (ResponseHandler pingHandler in pingHandlers.Values)
                    pingHandler.HandleResponse(response);
                pingHandlers.Clear();
            }

八叶�?/a> 2011-12-09 16:47 发表评论

Asterisk Manager Interface C++ Interpretor for Linux [转]

Sun, 27 Nov 2011 10:54:00 GMT

Asterisk Manager Interface C++ Interpretor for Linux �Q?a href="http://advancedcodingconcepts.blogspot.com/2011/02/asterisk-manager-interface-c.html">http://advancedcodingconcepts.blogspot.com/2011/02/asterisk-manager-interface-c.html�Q?/h3>

Through the past number of months in my full-time developer position for an online company and it's sales office, we've migrated to a VOIP telephony platform based on Asterisk, Linux and our custom call centre management application.

In order to integrate an autodialer in to the application, I had originally written a script in PHP to read from two asterisk servers (local and offshore) to post information about whether an agent is on a call or not, and also inbound calls to the call centre application.

The PHP script took a matter of hours to set up initially, but lacked proper structure - specifically, it could not track information when we switched to call queues, and was not easily portable between asterisk versions (1.8 locally, 1.2 in offshore)

For the past week I have worked on the design and initial coding of an app in C++ that acts as a client to both servers, interprets the messages and creates internal structures. I'm releasing that code here for review, and to help other's save some time, as I haven't noticed any other software out there that could accomplish this elegantly. Currently, I am calling this application AMIflex based on the flexibility it provides to manage the Asterisk AMI protocol.

This source code and it's derivatives can not be sold, licensed or packaged with any commercial software without my explicit permission.

Download the source code here (.tar.bz2, 6kb)

Installation instructions:

unpack the file via "tar -xvjf" command
create /etc/amiflex directory, and add a "servers.conf" file in the following format

Server: server name (for display purposes only)

Host: ip address or host name of ami server

Username: AMI manager username

Secret: AMI manager password/secret

Optional fields:

RetryPeriod: number of seconds to wait between connection retries - ie if asterisk is rebooted (default: 300)

Port: port number (default 5038)

You can insert more than one server (connections will be maintained together) by entering the first server's details, placing a second new line after the last configration line for that server, and then entering the details for the second server (and so on)

How to make this useful

When a server connection is established, there's a AMI::RegisterAllEvents() call after authentication. Add in your own events, and manipulate the structures or log output

Example:

AMI::RegisterEventCB("Dial", &AMI::MyDialCallback)

void AMI::MyDialCallback(MSG *msg)

{

if (msg->Attr("SubEvent").value=="Begin")

cout << msg->Attr("Channel").value + " is dialing " +msg->Attr("Dialstring").value<

}

Current limitations

I have some well known limitations with this software - I'll be resolving them myself for internal use, but my next step is to integrate proprietary MySQL links in to the source code, so I made sure to release this first. Currently my task list with my employer is too long to not take some shortcuts :)

The channel list linked list is a static member of the channel class - this means that you run AMIflex with multiple servers in the configuration file, all of the channels for all servers will be parallel. If you have two servers and each server has the same extension, then there will be one device entry created and two channels open on that device, when each extension is busy.
The bridge event currently only registered the first channel's Bridge member to the second member, and vice versa - If you are monitoring channels that will be bridged to more than one other channel, this will have to be expanded to a proper many to many-style linked list.

/Files/byc/amiflex.0.1.tar.bz2.rar

八叶�?/a> 2011-11-27 18:54 发表评论

dahdi

Thu, 24 Nov 2011 03:37:00 GMT

摘要: # lspci –vvvv Network controller # yum install kernel-develreboot # cd /usr/src/dahdi-linux-complete-XX # cd linux# make # make install # cd tools# ./configure# make# make install# make config... 阅读全文

八叶�?/a> 2011-11-24 11:37 发表评论

Asterisk + Vtiger CRM 5.1 �?sh��)击拨�?实现来电(sh��)弹屏

Sat, 05 Nov 2011 06:57:00 GMT

http://www.ztmaker.com/read.php?tid-325.html

一、Asterisk 端配�|?br />
修改配置文�g"/etc/asterisk/manager_custom.conf"�Q�在其中��d��一个管理帐�?vtigercrm"�Q?br />
[vtigercrm]
secret = vtigercrm
deny=0.0.0.0/0.0.0.0
permit=192.168.1.3/255.255.255.0
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
然后重启 Asterisk�Q?br />
二、CRM 端配�|?br />
1、设�|�模�?br />
使用��理员登�?VtigerCRM"�Q�在菜单中选择"Settings->Module Manager"�q�入模块列表��面�?br />
在此��面中找�?PBX Manager"模块�Q�将其启用后�Q�点击配�|�图标对此模块进行配�|�。各配置��内容如下：(x��)

Asterisk server IP: 填写 Asterisk 服务器地址
Asterisk server port: 填写 Asterisk ��理端口�Q�默认�ؓ(f��) 5038
Asterisk username: 填写 Asterisk 帐号名称�Q�按之前的配�|�就填写�?vtigercrm"
Asterisk password: 填写 Asterisk 帐号密码�Q�按之前的配�|�就填写�?vtigercrm"
Asterisk Version: 选择 Asterisk 的版�?br />2、设�|�用户分�?br />
使用用户帐号��d��后，点击右上方的"My Preferences"�Q�在打开的页面中扑ֈ�"Asterisk Configuration"栏，�q�作如下配置�Q?br />
Asterisk Extension: 填写用户的分机号
Receive Incoming Calls: 选中此选项
3、启动客��L(f��ng)��E�序

使用 SSH ��d�� CRM 服务器，�q�入 CRM �pȝ��安装路径下的"cron/modules/PBXManager"目录�Q�然后运行以下命令：(x��)

# ./AsteriskClient.php
如果一切正常就�?x��)出现如下信息�?x��)

Date: 05-03-2010
Connecting to asterisk server.....
Connected successfully

Trying to login to asterisk
Logged in successfully to asterisk server
��x(ch��ng)��Q�就可以使用�?sh��)话拨打刚才讄��的用户分机号了(ji��n)，如果一切正常，��可以看到CRM��面的右下角��׃��(x��)出现一个来甉|��C�框。另外在菜单中打开"Tools->PBX Manager"�Q�在��面中也可以看到来电(sh��)记录�?br />
注：(x��)如果 Asterisk �?1.4 版的�Q�请��C��下位�|�下载修改过�?AsteriskClient.php"来替换原�pȝ��中的文�g�Q?br />http://danielaliaman.com/blog///index.php/2009/07/23/vtiger_pbx_manager_issues_only_first_cal?blog=2

http://hi.baidu.com/ahhui/blog/item/e275b419ca6b415343a9ad83.html

如何使用vtigercrm5.1实现点击拨号、来�?sh��)弹�?/div>

环境�Q�Elastix1.6
已经实现�?span>功能�Q?span>vtigercrm点击客户�?sh��)话拨号�Q�来�?sh��)在vtigercrm弹屏�?br />未解决的问题�Q�弹屏没有来�?sh��)号码�?br />
一、vtigercrm点击呼出配置�Ҏ(gu��)��Q?br />1、首先保证asterisk、vtigercrm都正�怋�用�?br />
2、修�?etc/asterisk/manager.conf文�g�Q�增加如下：(x��)

[vtigecrm]
secret = vtigecrm
deny=0.0.0.0/0.0.0.0
permit=0.0.0.0/0.0.0.0
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user

复制代码

以上代码中vtigercrm不是特定的，但要在vtigercrm中的pbx讄��要一致�?br />
3、设�|�vtigercrm�Q?br />讑֮�-模块��理-PBX Manager

Asterisk server IP �Q?92.168.0.15 �Q�这是astersik服务�?/span>的IP�Q?br />Asterisk server port �Q?038 �Q�默�?038�Q?br />Asterisk username �Q�vtigercrm
Asterisk password �Q�vtigercrm
Asterisk Version�Q?.4

复制代码·

4、设�|�vtigercrm中的个�h分机��P��(x��)
我的讑֮�
Asterisk Extension�Q?00 �Q�这是asterisk里已讑֮�分机��P��(j��)
Receive Incoming Calls�Q?�?

5、修�Ҏ(gu��)��口文�Ӟ��(x��)
/var/www/html/vtigercrm/modules/PBXManager/utils/AsteriskClass.php
��以下代码：(x��)

switch($typeCalled){
case "SIP":
$context = "local-extensions";
break;
case "PSTN":
$context = "from-inside";//"outbound-dialing";
break;
default:
$context = "default";
}

复制代码

更改为：(x��)

switch($typeCalled){
case "SIP":
$context = "local-extensions";
break;
case "PSTN":
$context = "from-internal";//"outbound-dialing";
break;
default:
$context = "from-internal";
}

复制代码

如果��利�Q�到�q�里��p��用vtigercrm点击拨号�?ji��n)，点击��L(f��ng)��后，你的分机�?x��)响铃，响铃分机提机后�?span>�pȝ��p��动将��L(f��ng)��拨出�?br />参考：(x��)http://wiki.vtiger.com/index.php/vtiger510:Module_Asterisk_Howto
二、来�?sh��)弹屏的讄��?gu��)��Q?br />�q�里需�?span>�q�行一个文�Ӟ��q�且�q�行后，不能�l�止�?br />�q�入�q�个目录�Q?var/www/html/vtigercrm/cron/modules/PBXManager
�q�行�q�个命��o(h��)�Q?

php AsteriskClient.php

复制代码

�q�行后，不能�l�止�Q�如�?span>服务器重启过�Q�还需要再�ơ运行�?br />
到这里，来电(sh��)弹屏的功能也实现�?ji��n)，不过来�?sh��)昄��问题没找到答案�?br />��阅vtigercrm�C�֌�论坛�Q�也未找到合适的解决办法�Q�更有说此方法比较好CPU资源�?br />

弹屏补丁

vtigercrm/include/js/asterisk.js
function _defAsteriskTimer(){
var asteriskTimer = null;
var ASTERISK_POLLTIME = 5000; //vtigercrm polls the asterisk server for incoming calls after every 3 seconds for now
var ASTERISK_INCOMING_DIV_TIMEOUT = 60;

notificationPopup.js
function ResetPopup(){
  popupDiv.innerHTML = "";
  popupDiv.style.height = "0px";
  popupDiv.style.display = "none";
  parentDiv.removeChild(popupDiv);
  if(parentDiv.children.length ==0){
   parentDiv.style.display = "none";
  }

}

vtigercrm/cron/modules/PBXManager/AsteriskClient.php

function asterisk_handleResponse2($mainresponse, $adb, $asterisk, $state) {
$appdata = $mainresponse['AppData'];

$uniqueid = $channel = $callerType = $extension = null;
$parseSuccess = false;

if(
$mainresponse['Event'] == 'Newexten' && (strstr($appdata, "__DIALED_NUMBER") || strstr($appdata, "EXTTOCALL"))
) {

$uniqueid = $mainresponse['Uniqueid'];

  $channel = $mainresponse['Channel'];
  $splits = explode('/', $channel);
  $callerType = $splits[0];

  $splits = explode('=', $appdata);
  $extension = $splits[1];

  $parseSuccess = true;
} else if($mainresponse['Event'] == 'OriginateResponse'){
  //if the event is OriginateResponse then its an outgoing call and set the flag to 1, so that AsteriskClient does not pick up as incoming call
  $uniqueid = $mainresponse['Uniqueid'];
  $adb->pquery("UPDATE vtiger_asteriskincomingevents set flag = 1 WHERE uid = ?", array($uniqueid));
}else if($mainresponse['Event']=='NewCallerid'){

$channel = $mainresponse['Channel'];

if(strncmp($channel,"DAHDI",5)==0){

$uniqueid = $mainresponse['Uniqueid'];

   if(!empty($mainresponse['CallerID'])) {
    $callerNumber = $mainresponse['CallerID'];
   }elseif(!empty($mainresponse['CallerIDNum'])) {
    $callerNumber = $mainresponse['CallerIDNum'];
   }

   $sql = "UPDATE vtiger_asteriskincomingevents set from_number=? WHERE uid=?";
   $adb->pquery($sql, array($callerNumber, $uniqueid));
  }
}

八叶�?/a> 2011-11-05 14:57 发表评论

CentOS6.0 asterisk-1.6.2.20 freepbx-2.9.0 安装�q�程

Thu, 15 Sep 2011 10:15:00 GMT

1 安装CentOS6
需要安装的�?nbsp;
   Mysql数据库客服端�Q�Mysql数据库服务端
   PHP支持�Q�可选包增加��N��）(j��)
      php_mysql
   开发工�?br />
2 安装Asterisk
   yum install libxml2-devel
   yum install ncurses-devel

   groupadd asterisk
   useradd -c "asterisk PBX" -d /var/lib/asterisk -g asterisk -s /sbin/nologin asterisk

   解压asterisk
   ./configure
   make
   make install
   //make samples 安装f(xi��)reepbx不要�q�行
   make config

3 安装FreePBX

   解压freepbx

   yum install php-db
   //yum install php-pear-DB (centos 5)
   pear install db
   yum install php-posix

   service mysqld start

   mysqladmin create asterisk
   mysqladmin create asteriskcdrdb
   mysql asterisk < SQL/newinstall.sql
   mysql asteriskcdrdb < SQL/cdr_mysql_table.sql
   mysqladmin -u root -p password 123456

    /etc/httpd/conf/httpd.conf User apache 改�ؓ(f��) User asterisk ;Group apache 改�ؓ(f��) Group asterisk
   /etc/php.ini date.timezone = PRC
    /etc/sysconfig/selinux SELINUX=disabled

    setenforce 0

    service httpd start
    service asterisk start

    ./install_amp --username root --password 123456
    amportal start
    http://你的IP user:admin pass:admin

Asterisk Recordingvi/etc/amportal.conf
ARI_ADMIN_PASSWORD=你的密码
echo "/usr/local/sbin/amportal start">>/etc/rc.local

备注:
昄��PHP错误信息
调试的时候把php.ini中的display_errors   =   Off�Ҏ(gu��)��On
或�?nbsp; error_log   =   D:\Web\error.log   在error.log中查看错误日�?

FreePBX 2.X版本安装�?都是不��用密码就可以�q�入admin下去��理asterisk�Q�而有��x(ch��ng)��改密码的说明文档
官方�|�关没提供，论坛中也问得相对较多�Q�我��q��单说一下如何去配置FreePBX密码�?/p>

1、找�?etc/amportal.conf配置文�g�Q�将以下�q�个语句

# AUTHTYPE: authentication type to use for web admin
# If type set to 'database', the primary AMP admin credentials will be the AMPDBUSER/AMPDBPASS above
# valid: none, database
AUTHTYPE=none

更改为：(x��)�Q�注意红色标识）(j��)

# AUTHTYPE: authentication type to use for web admin
# If type set to 'database', the primary AMP admin credentials will be the AMPDBUSER/AMPDBPASS above
# valid: none, database
AUTHTYPE=database

2、执�?strong>./usr/src/freepbx-2.5.1/apply_conf.sh 使更改生效。（注意蓝色语句的完整性）(j��)

3、这样就可以使用

AMPDBUSER=
AMPDBPASS=

对应的帐号密码进入管理系�l�，也可以��?strong>Administrators模块建立�q�入�pȝ��的用戗��?/p>

报表不能查看

如果查看报表的时候出现如下的错误提示�Q?/span>

YOU MUST ACCESS THE CDR THROUGH THE ASTERISK MANAGEMENT PORTAL!

发生�q�个错误的原因可能是httpd�q�程不能��d��php�?x��)话的保存�\径�?/span>

1.执行以下命��o(h��)查看PHP�?x��)话的保存�\�?/span>:

grep save_path /etc/php.ini

扑ֈ�:

session.save_path = /var/lib/php/session

2. 修改权限

chown asterisk /var/lib/php/session

chmod -R 777 /var/lib/php/session

Couldn't load variables.txt

如果出现此问题可能是因�ؓ(f��)op_server.pl没有�q�行。进行目录进行即�?/span>

[root@www ~]# cd /var/www/html/panel/

[root@www panel]# ./op_server.pl

讄��freepbx 配置device与user分开

vi /etc/amportal.conf

AMPEXTENSIONS=deviceanduser | extensions

http://hi.baidu.com/kinnsei/blog/item/302c2d1e4f6952f01ad57601.html(��用�?user)与设�?device)区分开�?-内线分机的高�U�应�?

八叶�?/a> 2011-09-15 18:15 发表评论

久久久久这里只有精品,97精品伊人久久大香线蕉app,久久综合久久久

freepbc 呼出�{�待旉���修改

proxmox ve 2.2 elastix2.3模板制作

�?a2billing

http://www.osslab.com.tw/VoIP/IP_PBX/%e8%bb%9f%e9%ab%94%e5%bc%8f_IP_PBX/Asterisk_-_%e5%85%8d%e8%b2%bb_IP_PBX_%e7%b6%b2%e8%b7%af%e9%9b%bb%e8%a9%b1%e4%ba%a4%e6%8f%9b%e5%b9%b3%e5%8f%b0/Addons/A2Billing_%e7%ae%a1%e7%90%86%e7%af%87

前言

應用一�Q�國際電話卡模式

應用二：(x��)DID 轉接業務

應用三：(x��)預付制、月付制會員模式

應用五：(x��)Caller ID 認證模式

F.A.Q

Q:如何大量刪除 Customers?

Q:如何大量刪除 VoIP Settings?

Q:如何變更�pȝ��(d��ng)預設�q�別 USD

Q:匯入 ratecard 時總是出��N���?ERROR: file type is not allowed: application/force-download

盔R��連結

發現 Bug

asterisk 调试 讄���

Logging

Verbosity

Getting to Grips with sip.conf

星网锐捷 IDA 讑֤� 查看IP

elastix2.3 dashboard 无交换空间时 不显�C?bug 补丁

elastix2.3 中文语言�?voip88发布的elastix2.3中文版中抽取出来�?

elastix 2.3 升��到php5.2 CDR Report 出错 patch

elastix (Monitoring -> listen )ie 听录�?补丁

freepbx-mysql密码修改

asterisk-cpp

asterisk-dotnet 1.6.3.1 Asterisk 1.6.2.21 一直重复连�? BUG

Asterisk Manager Interface C++ Interpretor for Linux [转]

dahdi

Asterisk + Vtiger CRM 5.1 �?sh��)击拨�?实现来电(sh��)弹屏

CentOS6.0 asterisk-1.6.2.20 freepbx-2.9.0 安装�q�程

报表不能查看

Couldn't load variables.txt

freepbc 呼出�{�待旉��修改

Q:匯入 ratecard 時總是出��N��?ERROR: file type is not allowed: application/force-download

asterisk 调试讄��

elastix2.3 dashboard 无交换空间时不显�C?bug 补丁