Pulse DuMV@PCI Manual de usuario Pagina 1

DuMV@PCI 2 ports GSM/VoIP PCI Card User Manual PORTech Communications Inc.

-6- 8.System Information. 8.1 When you login the web page, you can see the demo system current system information like firmware version, company… et

-7- The DuMV@PCI will transfer to the URL according to the caller ID of the Mobile. *CID： (1) may enter the whole number, e.g. 0911111111 (2) onl

-8- *URL：The IP address to transfer this call (1) may enter the whole IP address, e.g. 192.168.0.101 or proxy extension or phone number. (

-9- *The call will be answered and prompt dial tone again. When the caller may enter the “Num”, system will connect the “URL” as destination. E.g

-10- 9.3 Call Back Service (50 sets) You can set call back service as the following steps (1) CID : set the phone number here (up to 50 sets) (2)

-11- 9.4 LAN to Mobile Settings The operator may assign 50 sets of routing rule to transfer the call incoming from LAN to MOBILE. The DuMV@PCI w

-12- *Call Num： 1.may enter the whole number, e.g. 0911111111 2.a simple *”means 2-stages-dialing. The call will be answered and prompt

-13- 10.Mobile 10.1 Mobile Status (1)Network Registration：The telecom carrier which the SIM card been registered. (2)SIM Card ID：SIM card ID. (3)S

-14- 10.2 Mobile Setting (1) VoIP Tx Gain: To adjust the volume of LAN side. GSM VoIP Codec LAN (6)Rx (5) Tx DTMF (1)VoIP Tx Gain (2) VoI

-15- (2) VoIP Rx Gain: To adjust the volume of Mobile side. (3)LAN Dialtone Gain: DTMF Reciver is not good,you can adjust gain down. (4) ON/Off: I

【Content】 1.INTRODUCTION... 1 2.FUNCTI

-16- (8)Presentation CLIR : If you need to block the Caller Id for call termination,please choose Suppression (9)Mobile PIN Code:If you

-17- So please, mark "Forward Enable" this blank to motivate this function. Take SJ Phone for example: Profiles -> Edit -> Advanced -

-18- 10.4 Mobile / SMS Agent : (1) Rx List: Read received SMS (2) Dest Num: the Receiver’s phone number (3) Message: Please fill the message that

-19- Click the serial no,you can view message as follows. 10.5 use AT Command via Telnet or your program Allows your program or Telnet Send/receive

-20- 11.Network In Network you can check the Network status, configure the WLAN Settings , LAN Setting and SNTP settings. 11.1 Network St

-21- 11.2 WAN Settings: You can check the current Network setting in this page. (1) The TCP/IP Configuration item is to setup the WAN port’s network

-22- 11.3 LAN Settings: You can check the current Network setting in this page. (1) The TCP/IP Configuration item is to setup the WAN port’s ne

-23- 11.4 SNTP Settings: SNTP Setting function: you can setup the primary and second SNTP Server IP Address, to get the date/time informa

-24- 12.SIP Setting In SIP Setting you can setup the Service Domain,Port Settings,Codec Settings,RTP setting,RPort Setting and Other SettingS. If th

-25- Example: Register VoipBuster Your Voipbuster username Your Voipbuster password Proxy Server’s IP

16.UPDATE ... 36 17.REBOOT..

-26- 12.2 Port Setting You can setup the SIP and RTP port number in this page. Each ISP provider will have different SIP/RTPport setting, please ref

-27- 12.3 Codec Settings: You can setup the Codec priority, RTP packet length in this page. You need to follow the ISP suggestion to setup these ite

-28- 12.4 Codec ID Setting You can setup the Codec ID in this page.

-29- 12.5 DTMF Setting You can setup the DTMF Setting in this page.

-30- 12.6 RPort Function: You can setup the RPort Enable/Disable in this page. To change this setting, please following your ISP information. When

-31- 12.7 SIP Responses 12.7.1 486(busy here), 503(Service unavailable): When Device is busy, you can select 486 or 505 to response to SIP. 12.7.2

-32- 12.8 Other Settings Other Settings: you can setup the Hold by RFC and QoS in this page. To change these settings. please following your ISP inf

-33- 13. NAT Trans In NAT Trans. you can setup STUN and uPnP function. These functions can help your VoIP device working properly behind NAT. 13.1

-34- 14.System Auth. In System Authority you can change your login name and password.

-35- 15.Save Change In Save Change you can save the changes you have done. If you want to use new setting in the VoIP system, You have to click the S

-36- 16.Update In Update you can update the system’s firmware to the new one or do the factory reset to let the system back to default setting. 16.

-37- 16.2 Restore Default Settings Default Setting you can restore the system to factory default in this page. You can just click the Restore button

-38- 17.Reboot Reboot function you can restart the system. If you want to restart the system, you can just click the Reboor button, then the system

-39- 18.Specification 18.1 Protocols SIP (RFC2543,RFC3261) 18.2 TCP/IP IP/TCP/UDP/RTP/RTCP/ CMP/ARP/RARP/SNTP DHCP/DNS Client IEEE802.1P/Q ToS/DiffSe

-40- CNG AEC, LEC Packet loss 18.5 GSM (DuMV@PCI) Dual BAND: 900/1800 MHZ Tri BAND(BenQ M23): 900/1800/1900 MHZ Tri BAND(Siemens MC56): 850/1800/1

-41- configure the box. Here are some screen shots showing all the important parameters. You have to note that in all the configuration process, th

-42- The mobile number you give in that page are the authorised mobile which can call GSM to Asterisk. These mobile number must be defined as you

-43- Once Asterisk configuration is made, you should get 'Registered' on the Realm1.

-44- It is very important to use only u-law or a-law as all DTMF is inband. So if you want to be able to do some DISA when you call from GSM to Ast

-45- username=103 fromuser=103 regexten=103 ; When they register, create extension 401 secret=xxxxxxx ; Asterisk extension password context=gatew

-1- 1.Introduction DuMV@PCI is a 2 channels VoIP GSM Gateway for call termination (VoIP to GSM ) and origination (GSM to VoIP). It is SIP based and

-46- 20.How to setup Asterisk to receive Caller ID from DuMV@PCI Test version trixbox-2.2 SIP Softphone  SJPhone 1.60.289a  X-Lite 1105x Modify

-47- nat=yes host=dynamic canreinvite=no context=internal  Add the following setting to /etc/asterisk/extensions.conf [internal] exten => 1000,1

-48- test1 pstn  call 0928492911(mobile number)  DuMV@PCI  hear the second dial tone,call SoftPhone’s number  SoftPhone  show pstn caller id

-49- Content-Type: application/sdp Content-Length: 242 v=0 o=root 2737 2737 IN IP4 192.168.66.202 s=session c=IN IP4 192.168.66.202 t=0 0 m=audio 15

-50- a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv test 2 SoftPhone  call 1002  DuMV@PCI

-51- c=IN IP4 192.168.66.145 t=0 0 m=audio 8000 RTP/AVP 0 8 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephon

-52- a=fmtp:101 0-16 a=silenceSupp:off - - - - register issue The packet date from Asterisk as follows. Please note, user_1002’s display name don’t

-53- eived=192.168.66.203;rport=5060 From: <sip:[email protected]>;tag=4e36d8f1 To: <sip:[email protected]> Call-ID: 7e45b773130f1fc9

-54- Via: SIP/2.0/UDP 192.168.66.203:5060;rport;branch=z9hG4bK672fa67f59c2223275f5ee286d27597a From: <sip:[email protected]>;tag=4e36d8f1 To

-55- OPTIONS sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.66.202:5060;branch=z9hG4bK7b92dd8a;rport From: "Unknown" <sip

-2- 4.Dimension: 13cm x 32.5cm (1) (3) (2)

-56- 21. Simple Steps Step 1. Change the Network setting if you need (Network/network setting) Step 2. Register SIP proxy Server or Asteri

-57- (2) *, specific mobile number when lan phone call in, DuMV@PCI will connect with the specific mobile number auto. (3) *,#--->It is 1 stage

-3- 5.Chart of the device 5.1 Antenna：Antenna connector. 5.2 SIM Slot 2: Insert second SIM card 5.3 SIM Slot 1: Insert first SIM card 5.4

-4- 6.CABLING 6.1 Connect the internet cable from HUB to the ‘WAN’ connector of the DuMV@PCI. *If you need to stack up more DuMV@PCI, y

-5- 7.Web Page Setting When the IP setting is done, the operator may setup all the rest parameters via web page. Browse the IP add