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
2) under bristuff, run the command: ./install.sh
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:
=============================
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:
=========================
[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:
========================
[from-gsm] // call your SIM number will forward to a test number. 10000 for telcom service in china.
exten=> s, 1, Answer();
exten=> s, n, Dial(zap/5/10000); // call to china telcom service
[from-internal] // from sip dial to the test number
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
2) modprobe ztgsm
3) ztcfg -vvvvvv
4) demsg:
=====================
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:
==============================
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:
================================================================
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:
===============================
*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:
[email protected]: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.
===============================
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.
================================
to get the latest info, please refer wiki:
wiki.openvox.cn/index.php/G400P
regards!
James.zhu