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电梯直达

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发表于 2009-07-02 17:02 |只看该作者 |倒序浏览

hello,
we will release openvox G400P very soon. here , i posted a test version. please go though these steps， if you have a test card:
1) unzip the bristuff to /usr/src // 解压 bristuff
2) under bristuff, run the command: ./install.sh // 运行安装脚本
3) make sure all steps are done correctly without any errors.
4) replace the patch files, which you can get from openvox website.
5) under bristuff directory, run the command: ./compile.sh
6) edit the zaptel.conf like this: // 编译 zaptel.conf 文件
=============================
loadzone=nl
defaultzone=nl
alaw=1,3,5,7
span=1,1,3,ccs,ami
span=2,2,3,ccs,ami
span=3,3,3,ccs,ami
span=4,4,3,ccs,ami
bchan=1,3,5,7
dchan=2,4,6,8
================
7) edit the zapata.conf like this: // 编译 zapata.conf 文件
=========================
[channels]
txgain = 0.0
rxgain = 0.0

signalling = gsm
context = from-gsm
echocancel=no
relaxdtmf=yes
; slot A
channel => 1
; slot B
channel=> 3
; slot C
channel => 5
; slot D
channel => 7
=================

edit the dialplan like this:  // 拨号规则
========================
[from-gsm]  //  呼入以后会自动打到中国电信的10000 号，测试了。啊啊。对不起。政府
exten=> s, 1, Answer();
exten=> s, n, Dial(zap/5/10000); // call to china telcom service

[from-internal]  // 注册一个sip 电话，从 sip 呼出到 10000 号
exten => 100,1,Dial(zap/1/10000)
exten => 100,2,Hangup

exten => 200,1,Dial(zap/3/10000)
exten => 200,2,Hangup

exten => 300,1,Dial(zap/5/10000)
exten => 300,2,Hangup

exten => 400,1,Dial(zap/7/10000)
exten => 400,2,Hangup
========================
until here, you have set all necessary files, please run the driver and asterisk in the way:
1) modprobe zaptel // 加载 zaptel
2) modprobe ztgsm // 加载 ztgsm 模块
3) ztcfg -vvvvvv  // 启动通道
4) demsg:  // 显示驱动的启动信息
=====================
Zapata Telephony Interface Unloaded
Zapata Telephony Interface Registered on major 196
Zaptel Version: 1.2.27
Zaptel Echo Canceller: MG2
ztgsm: no version for "zt_receive" found: kernel tainted.
ACPI: PCI Interrupt 0000:02:0c.0[A] -> GSI 20 (level, low) -> IRQ 217
ovgsm: iomem at feae0000 size 65536
ovgsm: iomem remote to f8be0000
ovgsm: OpenVox G400P card configured at IRQ 217 io mem f8be0000 HZ 1000
ovgsm: slot 0 is Installed
ovgsm: slot 1 is Installed
ovgsm: slot 2 is Installed
ovgsm: slot 3 is Installed
ovgsm: module 0 status is ON
ovgsm: module 1 status is ON
ovgsm: module 2 status is ON
ovgsm: module 3 status is ON
ovgsm: Powering up all spans... done.
pci_io addr is 0xf8be0000
ovgsm: adcgain is 0x27(-3db)
ovgsm: dacgain is 0x5b(-15db)
ovgsm: sidetone set to mute
ovgsm: adcgain is 0x27(-3db)
ovgsm: dacgain is 0x5b(-15db)
ovgsm: sidetone set to mute
ovgsm: adcgain is 0x27(-3db)
ovgsm: dacgain is 0x5b(-15db)
ovgsm: sidetone set to mute
ovgsm: adcgain is 0x27(-3db)
ovgsm: dacgain is 0x5b(-15db)
ovgsm: sidetone set to mute
ovgsm: 1 OpenVox G4XX card(s) in this box, 4 GSM spans total.
Registered tone zone 3 (Netherlands)
==========================
5) asterisk -vvvgc ; make sure the gsm is up and connected with network.
asterisk console will show some messages:
=========GSM 卡被激活=====================
Asterisk Ready.
*CLI>    -- GSM Span 4 registered to network!
-- GSM Span 1 registered to network!
-- GSM Span 2 registered to network!
-- GSM Span 3 registered to network!
=====================================
6) test sms from asterisk console:
under asterisk console, run:
==============通过asterisk sms 发短信到你的手机==================================================
gsm send sms 1 1357078XX "hello, this is openvox gsm card."
================================================================
here, using channel 1 to send a sms to your number. you will receive a message from asterisk server.
7) test a outbound call：
===========呼出的例子====================
*CLI> 2009-07-02 15:01:21 DEBUG[3175]: chan_sip.c:7590 check_user_full: Setting NAT on RTP to 524288
2009-07-02 15:01:21 DEBUG[3175]: chan_sip.c:1449 __sip_ack: Stopping retransmission on 'NDRhYjdjYmUxOTkxMThlZTg4NzJlYmQwOWJkZmU2Njg.' of Response 1: Match Found
2009-07-02 15:01:21 DEBUG[3175]: chan_sip.c:7590 check_user_full: Setting NAT on RTP to 524288
2009-07-02 15:01:21 DEBUG[3175]: chan_sip.c:11069 handle_request_invite: Checking SIP call limits for device 100
2009-07-02 15:01:21 DEBUG[3175]: chan_sip.c:6490 build_route: build_route: Contact hop: <sip:100@192.168.2.179:32356>
-- Executing Dial("SIP/100-09908bf0", "zap/1/10000"

in new stack
-- Called 1/10000
2009-07-02 15:01:21 DEBUG[3199]: chan_sip.c:3167 sip_rtp_read: Oooh, format changed to 4
2009-07-02 15:01:29 DEBUG[3175]: chan_sip.c:1449 __sip_ack: Stopping retransmission on 'NDRhYjdjYmUxOTkxMThlZTg4NzJlYmQwOWJkZmU2Njg.' of Response 2: Match Found
2009-07-02 15:01:29 DEBUG[3199]: chan_zap.c:2739 zt_hangup: Hangup: channel: 1 index = 0, normal = 20, callwait = -1, thirdcall = -1
2009-07-02 15:01:29 DEBUG[3199]: chan_zap.c:3259 zt_setoption: Set option TDD MODE, value: OFF(0) on Zap/1-1
2009-07-02 15:01:29 DEBUG[3199]: chan_zap.c:1726 update_conf: Updated conferencing on 1, with 0 conference users
-- Hungup 'Zap/1-1'
=========================================

this is a very brief doc for G400P, we will later release a full doc for G400P.
=========测试环境======================
Test tools:
1) centos-5.3 with kernel- 2.6.18-128.el5 #
2) bristuff-0.3.0-PRE-1y-u with openvox patch
3) SIM cards with the pin service disabled.
================================
regards!
James.zhu