MV-370 / MV-372 VoIP GSM Gateway User Manual
Contents
1. gt it is two stage dialing when lan phone call in MV 370 MV 372 will provide dial tone and you can enter mobile number 2 specific mobile number when lan phone call in MV 370 MV 372 will connect with the specific mobile number auto 3 gt It is 1 stage dialing When lan phone and MV 370 MV 372 both register Asterisk you can dial any destination number from lan phone directly Please note Asterisk need to set route of destination number that dial out from MV 370 MV 372 All changes both need to click save and change 60 15 21 Federal Communications Commission FCC Statement You are cautioned that changes or modifications not expressly approved by the part responsible for compliance could void the user s authority to operate the equipment 15 105 b Federal Communications Commission FCC Statement This equipment has been tested and found to comply with the limits for a Class B digital device pursuant to part 15 of the FCC rules These limits are designed to provide reasonable protection against harmful interference in a residential installation This equipment generates uses and can radiate radio frequency energy and if not installed and used in accordance with the instructions may cause harmful interference to radio communications However there is no guarantee that interference will not occur in a particular installation If this equipment does cause harmful interference to radio or tel2. SMS Reader 16 25 27 08 03 12 896935366862 Mw Serial can send SMS and receive SMS Back Delete 11 5 use AT Command via Telnet or your program Allows your program or Telnet Send receive SMS with AT Command Port 23 username voip password user level 1 command logout module modulel module2 gt modulel getting module 1 Choose module got press ctrl x to release module 1 Enter ate1 then you can see your at command below Please enter account and password ate1 0 at cmgf 1 gt test gt 21 12 Network In Network you can check the Network status configure the WLAN Settings LAN Setting and SNTP settings 12 1 Network Status You can check the current Network setting in this page PO RTech Network Status Your CTI Partner nous Mobile Type Fixed IP Client Fixed IP Client IP 192 168 0 122 192 168 0 102 Network Mask 255 255 255 0 255 255 255 0 Gateway 192 168 0 254 192 168 0 254 WAN Safina MAC 00037 E009999 00037 E008888 12 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 environment You may refer to your current network environment to configure the system properly 2 The PPPoE Configuration item is to setup the PPPoE Username and Password If you have the PPPoE account from your Service Provider please inpu
3. Route Mobile Reboot system Network SIP Settings NAT Transform Update System Authority Save Change 40 19 IP Setting The operator can setup or query the network parameters by dialing in the mobile number which it SIM card has been put in the main body The status or result is response by voice In the first 20 seconds after power on the VoIP GSM Gateway enters the IP setting mode The operator may dial in the mobile number during this period to set or query Notes After you hear Option Successful hang up Unit will reboot automatically System will automatically Reboot WARNING ALL User Changeable NONDEFAULT SETTINGS WILL BE LOST This will include network and service provider data IVR will announce the current IP address Default 192 168 0 100 IVR will announce if DHCP in enabled or disabled default OFF IVR will announce the current network mask Default 255 255 255 0 IVR will announce the current gateway IP address Default 192 168 0 254 IVR will announce the current IVR Menu Choice 195 1 98 1 20 121 1 23 1244 1 25 4 the network parameters IVR Action Reboot Factory Reset Check IP Address Check IP Type Check Network Mask Check Gateway IP Address Check Primary Item 1 setting in the Primary DNS field Default 192 168 0 1 IVR will announce the
4. O A 22 ISP SEN ea GORR 26 PA MAM TERA NO ae a da Aa k rab o eb ea ete ae es o be i 35 SYSTEM U Z ee ee e fe tese ende ANINA A Pen ma 36 VO SAVE CHANGE si SN AA aaa A bs titus edd ad ae or AERE 37 A a dm DUK tse E 38 RR DOO V ioni med ise a etu Le a Sd edo Mage 40 I9 TP SE TUN Gn 4 ZO SPEC TETAS TION td o tence ber atum 43 21 APPENDIX SETUP MV 370 MV 372 WITH ASTERISK eee 44 22 HOW TO SETUP ASTERISK TO RECEIVE CALLER ID FROM MV 370 MV 372 50 23 SIMPLE STEPS ta iai de ideis tte bee od evista eae 60 1 Introduction MV 370 MV 372 is a 1 2 channels VoIP GSM Gateway for call termination VoIP to GSM and origination GSM to VoIP It is SIP based and compatible with Asterisk It can enable to make 1 2 calls simultaneously from IP phones to GSM networks and GSM network to IP phone 2 Function description 2 1 VolP SIP gt GSM MV 370 MV 372 conversion 2 2 50 sets of LAN gt MOBILE routes setting gt 50 sets of MOBILE gt LAN routes setting 2 3 Voice response for setting and status dial in from mobile 2 4 Series connections to save bills 2 5 Standard SIP RFC2543 RFC3261 protocol Communicates with other gateway or PC 3 Parts list Please check the parts for any missing parts If do please contact our agents 3 1 MV 370 MV 372 main body 3 2 Power adaptor AC DC 110V AC 12V DC or
5. Save Change Reboot O O 4 C C1 BP C O The MV 370 MV 372 will transfer to the mobile number according to the incoming URL URL The IP address of the incoming call may enter the whole IP address e g 192 168 0 101 or proxy server s extension If a simple is entered means no restriction for the incoming IP address 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 dial tone again to receive the called number as the 12 destination e g 0911111111 or 09111111112 3 d n a ppp for one stage dialing is option d n means to delete the beginning n codes a ppp means to add ppp in front for example d2a09 means one stage dialing delete the first 2 codes from your destination number then add 09 in front as the new destination number Example Lan to Mobile 1 MV 370 MV 372 and Lan Phone both need to register proxy server or Asterisk 2 Proxy server asterisk set the route that the prefix of destination number 3 When you dial any destination phone number from lan phone MV 370 MV 372 will connect this call auto Example of Application When you call the ch 1 MV 370 MV 372 gsm number it will provide dial tone and you enter a destination number Then ch 2 MV 370 MV 372 will dial this number and connect ch 1 MV 370 MV 372 mobile to lan set route table ch 2 MV 370 MV 372 lan to mobile set route tabl
6. System Authority Save Change Reboot 13 6 RPort Function You can setup the RPort Enable Disable in this page To change this setting please following your ISP information When you finished the setting please click the Submit button PORTech RPort Setting Your CTI Partner Route RPort of Mobile1 On O Of Mobile RPort of Mobile 2 90n O Off Network Gina Resst SIP Settings Service Domain Port Settings Codec Settings Codec ID Setting DTMF Setting RPort Setting SIP Responses Other Settings NAT Transform Update System Authority Save Change Reboot 32 13 7 SIP Responses PO RTech SIP Responses Setting CTI Ada Route Response on port busy Mobile O 486 Busy here Network 0 503 Service unavailable SIP Settings z SIP Responses Service Domain SON OOFF 180 Ringing Auto force to ON if 183 was OFF Port Settings OON GOFF 183 Session Progress Codec Settings Codec ID Setting AE nu sip Saar er Settings NAT Transform Update System Authority Save Change Reboot 13 7 1 486 busy here 503 Service unavailable When Device are busying you can select 486 or 505 to response to SIP 13 7 2 180 Ring on off LAN TO MOBILE two stage dialing can be turn off therefore there will be no the Ring Back Tone all the phone call will be transferred to Voice Mail directly For this function 183 must be turn on 13 7 3 183 Session Progress
7. 192 168 66 202 gt tag as2a2fbf98 Call ID F4B32CA6 1835 4E68 941A C685B39C43FF 192 168 66 145 CSeq 63148 INVITE User Agent Asterisk PBX Allow INVITE ACK CANCEL OPTIONS BYE REFER SUBSCRIBE NOTIFY Contact lt sip 1002 192 168 66 202 gt 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 13798 RTP AVP 0 8 101 a rtpmap 0 PCMU 8000 a rtpmap 8 PCMA 8000 a rtpmap 101 telephone event 8000 55 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 appear So the website s Display Name is not available lt SIP read from 192 168 66 203 5060 REGISTER sip 192 168 66 202 SIP 2 0 Via SIP 2 0 UDP 192 168 66 203 5060 rport branch z9hG4bK590e92b551 233a10a0ae71944c19b5 aa From lt sip 1002 192 168 66 202 gt tag 4e36d8f1 To lt sip 1002 192 168 66 202 gt Call ID 7e45b773130f1fc945efcee502f840420192 168 66 203 Contact lt sip 1002 192 168 66 203 5060 gt CSeq 10 REGISTER Expires 300 Authorization Digest username 1002 realm asterisk nonce 3ca93a1e response 4d39ccb0dae64 bb2f1341e9896ac1ea7 uri sip 192 168 66 202 algorithm MD5 User Agent CMI CM5K Content Length 0 11 headers 0 lines Using latest REGISTER request as basis request Sending to 192 168 66 203 5060 NAT Transmitting N
8. If you need to register to Asterisk and proxy server please choose this option And how to transfer the caller ID to LAN please refer 21 How to setup Asterisk to receive Caller ID from MV 370 MV 372 page 42 MV 370 MV 372 will send the message as follows in the Packet From caller number lt sip 3001 192 168 0 228 gt tag 51088abb 0 Tel Tel MV 370 MV 372 will send the message as follows in the Packet From caller number sip caller number 2192 168 0 228 gt tag 6ac93f7c lt Please note lf you choose this option please don t register to Asterisk and proxy server Please only fill proxy server ip and choose Active on else field empty in sip setting service demain User Tel MV 370 MV 372 will send the message as follows in the Packet From Username sip caller number 192 168 0 228 gt tag 7f130947 X If you choose this option please don t register to Asterisk and 16 proxy server Please only fill proxy server ip Username and choose Active on else field empty in sip setting service demain 9 Presentation CLIR If you need to block the Caller Id for call termination please choose Suppression 10 Mobile PIN Code lf you need to unlock pin code via MV 370 MV 372 you can click On and enter pin code 11 LAN Answer Mode Answered when mobile answer then connect the call Alerted when the mobile is ringing back tone then connect the call
9. LAN Settings SNTP Settings SIP Settings NAT Transform Update System Authority Save Change Reboot 12 4 SNTP Settings SNTP Setting function you can setup the primary and second SNTP Server IP Address to get the date time information Also you can base on your location to set the Time Zone and how long need to synchronize again When you finished the setting please click the Submit button 0181661 swrp Settings Your CTI Partner You could set the SNTP servers in this page Route Mobile SNTP 0n Oof Network o Primary Server time windows cam WAN Settings Secondary Server 208 184 49 8 LAN Settings SNTF Settings Time Zone GMT v 08 v 00 Y hh mm SIP Settings Sync Time 1 0 0 dd hh mm NAT Transform Update System Authority Save Change Reboot 195 13 SIP Setting In SIP Setting you can setup the Service Domain Port Settings Codec Settings RTP setting RPort Setting and Other SettingS If the VoIP service is provided by ISP you need to setup the related informations correctly then you can register to SIP Proxy Server correctly 13 1 In Servcie Domain Function you need to input the account and the related informations in this page please refer to your ISP Provider You can register three SIP accounts You can dial the VoIP phone to your friends via first enable SIP account and receive the phone from
10. a rtpmap 0 pcmu 8000 53 a rtpmap 8 00 a rtpmap 3 gsm 8000 a rtpmap 101 telephone event 8000 a fmtp 101 0 15 a sendrecv test 2 SoftPhone gt call 1002 gt MV 370 MV 372 gt hear second dial tone and call pstn gt pstn answer gt show caller id mobile number 092849291 1 This Is X Lite receiving packet Test ok INVITE sip 1002 192 168 66 202 SIP 2 0 Via SIP 2 0 UDP 192 168 66 145 7331 rport branch 29hG4bK4C4315351FC84CA582D14FB8C25F C3BF From user_1001 lt sip 1001 192 168 66 202 7331 gt tag 1121869743 To lt sip 1002 192 168 66 202 gt Contact lt sip 1001 192 168 66 145 7331 gt Call ID F4B32CA6 1835 4E68 941A C685B39C43FF 192 168 66 145 CSeq 63148 INVITE Proxy Authorization Digest username 1001 realm asterisk nonce 0d3b2879 response 8aaaaa5b5ad53 654bf0a2ab0fa9bb1 18 uri sip 1002 192 168 66 202 algorithm MD5 Max Forwards 70 Content Type application sdp User Agent X Lite release 1105x Content Length 254 v 0 0 1001 5111461 5111501 IN IP4 192 168 66 145 s X Lite 54 c IN 456 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 telephone event 8000 a fmtp 101 0 15 a sendrecv SIP 2 0 200 OK Via SIP 2 0 UDP 192 168 66 145 7331 branch 29hG4bK4C4315351FC84CA582D14FB8C25FC3BF received 192 168 66 145 rport 7331 From user 1001 lt sip 1001 192 168 66 202 7331 gt tag 1121869743 To lt sip 1002
11. an entered it means 2 stages dialing The call will be answered and prompt dial tone again to receive the IP address sip extension or any phone number as the destination The caller may enter the IP such as 192 168 0 101 If the device have register proxy server Asterisk you can enter any destination phone number Please note the proxy server Asterisk need to set the route of destination phone number Example 1 Mobile to Lan 0932 0911123456 MV 370 MV 372 have register proxy server Asterisk The proxy server Asterisk have the route 09 When the callers prefix number is 0932 MV 370 MV 372 will connect 0911123456 automaticlly 2 Mobile to Lan Any caller call the MV 370 MV 372 s sim MV 370 MV 372 will prompt dial tone Caller can enter IP or sip extension or phone number sip extension or phone number both need to register SIP Proxy Server or Asterisk Phone number SIP Proxy Server or Asterisk need to set the route of this phone number 10 2 Call Back Service 50 sets New feature 00181661 vobile To LAN Table Your CTI Partner Route Page Y CID URL 3 0933579613 Lan To Mobile Settings 1885933579513 O Mobile Network SIP Settings NAT Transform Update System Authority Save Change Reboot 0 O C C BR U N o Delete Selected Delete All reset Add New Position 0 49 CID Ex 0911111111 0911 URL Ex 192 168 0 1 28t You can set call back service as the following step
12. 00 MHZ Tri BAND Siemens MC56 850 1800 1900 MHZ Quad BAND 900 1800 1900 850 MHZ 21 Appendix Setup MV 370 MV 372 with Asterisk 21 1 Usage A typical usage of such a gateway is to be able to give a call with your normal mobile to any destination at voip cost Your mobile lt gsm network gt MV 370 MV 372 lt lan gt Asterisk lt internet gt VOIP provider lt whatever gt landline To do such a call you just call your MV 370 MV 372 number it has its own simcard then you get an invitation tone then you dial the number which is handled by Asterisk If you have some special deals with your mobile operator like free special number you can call your MV 370 MV 372 for free You can then call all around the world from your mobile at voip cost 21 2 MV 370 MV 372 Configuration Once you ve configured everything in the box one good advice is to unplug the power and to restart it By this way you should have all the parameters taken into account To have the MV 370 MV 372 to work with Asterisk you need first to 44 configure the box Here are some screen shots showing all the important parameters You have to note that in all the configuration process the MV 370 MV 372 is considered as extension 103 of the IPBX In Bold are the parameters depending on your installation WAN Settings You could configure the WWAN settings in this page WAN Setting IP Type CO Fixed IP C DHCP Client C PPPoE I
13. 220V AC 12V DC 3 3 Network cable 3 4 Antenna 3 5 User Manual s Mme 3 1 MV 370 3 1 MV 372 2 3 4 oe MV 372 Mobile VolP CPORTe Your CTI pa 5 MV 370 Panel description Ja 14 5cm 5 1 5 2 5 3 5 4 5 6 5 5 5 7 5 1 Antenna Antenna connector 5 2 DC 12V Power socket 5 3 LAN Standard RJ 45 socket connecting to Hub circuit 5 4 PWR Power indicator light red light Light is on when system s power supply is normal 5 5 MOBILE GSM indicator light green light Light flashes when GSM status is normal light turns on constantly when GSM is called 5 6 LAN LAN indicator light green light Light flashes when Lan is called light turns off when GSM answered 5 7 LINK Link indicator light green light Light is on when network is connected correctly 6 MV 372 Panel description 6 1 PORT Your CT ES Tech 6 2 63 6 4 6 5 6 6 6 76 8 6 1 Antenna Antenna connector 6 2 DC 12V Power input 6 3 LAN LAN port It also can be DHCP Server 6 4 WAN RJ 45 internet connector standard RJ 45 socket connect to HUB 6 5 PWR Power LED Light up when power is normal 6 6 VoIP1 an indicator light of VoIP 1 6 7 VoIP2 an indicator light of VoIP2 6 8 LINK Indicator Light up when network is connected 7 CABLING 7 1 Connect the internet cable from HUB to the WAN connector of the MV 372 If you need to stack up more MV 372 you can stac
14. 3 ee URL Port Fwd to Mobile1 peoos Fwd to Mobile2 p migos Fwd to External o The Explanation of Picture Fwd to Mobile1 192 168 0 100 5060 it means when 5062 Port are busying SJ Phone can transfer the call to 5060 Port 192 168 0 100 Fwd to Mobile2 192 168 0 100 5062 it means when 5060 Port are busying SJ Phone can transfer the call to 5062 Port 192 168 0 100 e f both 5060 port and 5062 port are busying at same time you can set up Fwd to External then you can transfer the phone call to another designate device 9 11 4 Mobile SMS Agent PORTech SMS Agent Your CTI Partner Read received SMS Route Port Status Mobile 1 Standby Mobile Mobile 2 Standby Status SEE I SMS Sender SMS Agent Via Mobile 1 2 SPRE Dest Num Maximum Number of UCS2 chars for this text box is 70 SIP Settings NAT Transform Message Update zl System Authority You have 70 UCS2 chars remaining for your description Save Change Reboot 1 Rx List Read received SMS 2 Dest Num the Receiver s phone number 3 Message Please fill the message that want to send to receiver When you click Rx List you can view all received SMS as follows SMS Rx Last Read Status ETS a REC READ 886936114545 08 01 01 19 34 22 2 REC READ 885935385862 08 03 12 16 25 27 Click the serial no you can view message as follows 20
15. 3130f1fc945efcee502f840420192 168 66 203 Contact lt sip 1002 192 168 66 203 5060 gt CSeq 11 REGISTER Expires 300 Authorization Digest username 1002 realm asterisk nonce 5def9231 response 046a412f14e7ed4 e98fd507416994a80a uri sip 192 168 66 202 algorithm MD5 User Agent CMI CM5K Content Length 0 11 headers 0 lines Using latest REGISTER request as basis request Sending to 192 168 66 203 5060 NAT Transmitting NAT to 192 168 66 203 5060 SIP 2 0 100 Trying Via SIP 2 0 UDP 192 168 66 203 5060 branch z9hG4bK672fa67f59c2223275f5ee286d27597a recei ved 192 168 66 203 rport 5060 From lt sip 1002 192 168 66 202 gt tag 4e36d8f1 To lt sip 1002 192 168 66 202 gt Call ID 7e45b773130f1fc945efcee502f840420192 168 66 203 CSeq 11 REGISTER User Agent Asterisk PBX Allow INVITE ACK CANCEL OPTIONS BYE REFER SUBSCRIBE NOTIFY Contact lt sip 1002 192 168 66 202 gt Content Length 0 12 headers 0 lines Reliably Transmitting NAT to 192 168 66 203 5060 58 OPTIONS sip 1002 192 168 66 203 5060 SIP 2 0 Via SIP 2 0 UDP 192 168 66 202 5060 branch z9hG4bK7b92dd8a rport From Unknown lt sip Unknown 192 168 66 202 gt tag as5dee3942 To lt sip 1002 192 168 66 203 5060 gt Contact lt sip Unknown 192 168 66 202 gt Call ID 5ebc2211278e2cb769991 1ad39454d4e 192 168 66 202 CSeq 102 OPTIONS User Agent Asterisk PBX Max Forwards 70 Date Tue 22 May 2007 03 11 54 GMT Allow IN
16. 72 103 type friend 48 username 103 fromuser 103 regexten 103 When they register create extension 401 secret xxxxxxx Asterisk extension password context gateway Incoming calls context dtmfmode inband Very important for DISA to work call limit 1 Limit to 1 call max callerid GSM Gateway lt 103 gt host dynamic nat no Gateway is not behind a NAT router canreinvite no Typically set to NO if behind NAT insecure very qualify yes disallow all allow ulaw prefered codec for DTMF detection allow alaw 21 6 extensions conf FEE GSM Gateway incoming calls 7 gateway exten gt 103 1 Answer exten gt 103 2 DigitTimeout 3 give enough time to do second stage dialing exten gt _103 3 ResponseTimeout 5 exten gt 103 4 DISA no password outgoing here outgoing is the normal context to deal with the dial plan outgoing example of LAN to GSM call call the MV 370 MV 372 sim card mail box thru GSM exten gt _888 1 SetCallerID xXxxxxxxxxx exten gt _888 2 Dial SIP S EXTEN 103 60 r exten gt _888 3 Hangup 49 ID from Caller 22 How to setup Asterisk to receive MV 370 MV 372 est version trixbox 2 2 SIP Softphone e SJPhone 1 60 289a e X Lite 1105x Modify file e Add the following setting to etc asterisk sip conf 1000 type friend secret 1000 qualify yes nat yes host dynamic canreinvite no context internal 1001 type friend
17. AT to 192 168 66 203 5060 SIP 2 0 100 Trying Via SIP 2 0 UDP 192 168 66 203 5060 branch z9hG4bK590e92b551233a10a0ae71944c19b5aa rec 56 eived 192 168 66 203 rport 5060 From lt sip 1002 192 168 66 202 gt tag 4e36d8f1 To lt sip 1002 192 168 66 202 gt Call ID 7e45b773130f1fc945efcee502f840420192 168 66 203 CSeq 10 REGISTER User Agent Asterisk PBX Allow INVITE ACK CANCEL OPTIONS BYE REFER SUBSCRIBE NOTIFY Contact lt sip 1002 192 168 66 202 gt Content Length 0 Transmitting NAT to 192 168 66 203 5060 SIP 2 0 401 Unauthorized Via SIP 2 0 UDP 192 168 66 203 5060 branch z9hG4bK590e92b551233a10a0ae7 1944c19b5aa rec eived 192 168 66 203 rport 5060 From lt sip 1002 192 168 66 202 gt tag 4e36d8f1 To lt sip 1002 192 168 66 202 gt tag as13a32ae8 Call ID 7e45b773130f1fc945efcee502f840420192 168 66 203 CSeq 10 REGISTER User Agent Asterisk PBX Allow INVITE ACK CANCEL OPTIONS BYE REFER SUBSCRIBE NOTIFY WWW Authenticate Digest algorithm MD5 realm asterisk nonce 5def9231 Content Length 0 Scheduling destruction of call 7e45b773130f1fc945efcee502f84042 192 168 66 203 in 15000 ms asterisk1 CLI gt lt SIP read from 192 168 66 203 5060 REGISTER sip 192 168 66 202 SIP 2 0 57 Via SIP 2 0 UDP 192 168 66 203 5060 rport branch2 z9hG4bK672fa67159c2223275f5ee286d27597a From lt sip 1002 192 168 66 202 gt tag 4e36d8f1 To lt sip 1002 192 168 66 202 gt Call ID 7e45b77
18. Codec Priority 2 G 711 a law v Codec Priority 3 Mot Used v Codec Priority 4 Not Used v Codec Priority 5 Not Used vj Codec Priority 6 Not Used v Codec Priority Not Used v Codec Priority 8 Mot Used Ww 711 8 5 729 20 ms G 723 30 ms 7235 O On of Voice VAD O On Off 47 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 Asterisk it has to be one of these 2 codecs Mobile Setting VoIP Tx Gain HO 1 12 VoIP Rx Gain 3 0 15 LAN Dialtone Gain 10 0 12 Mobile ON OFF Routing Range 0 lta 49 0 49 CODEC Tx Gain 6 0 7 CODEC Rx Gain 6 7 SIP From Tel User Standard v Answer Delay 0 0 15 CLID Presentation Suppression Invocation These settings seem to be ok just adjust 21 3 Antenna position Another important thing is to properly place the provided antenna If your gsm reception is good you should get around 18 or 19 as Signal Quality in the Mobile Status page With that level of signal quality your audio quality will be very good On the other end the signal quality down to 11 audio becomes very jerky So maximum signal quality maximum audio quality 21 4 Asterisk configuration Once the MV 370 MV 372 is set you have to configure Asterisk On that side you have to setup files as follow 21 5 sip conf GSM VOIP Gateway MV 370 MV 3
19. Income when lan dial out then connect soon 12 Answer Delay Delay for incoming call when the ring 13 When you buy Quad band you need to choose your GSM frequency 11 3 Mobile Forward Setting When the first route are busying SIP can transfer phone call to another free route When the device are busying the phone call can be transfer to another device external equipments SEJE PO RTech Forward Setting Bb CTI tl Route Mobile A URL Port Status Fwd to 1 192 168 0 100 5060 Fwd to Mobile2 192 168 0 100 5062 Fwd Settings SMS Agent Network SIP Settings NAT Transform Update System Authority Save Change Reboot O Forward Enable Fwd to External Forward Enable is not motivate on Defualt value So please mark Forward Enable this blank to motivate this function Take SJ Phone for example Profiles gt Edit gt Advanced gt Accept redirection replies Turn on the Forward Enable therefore the SJ Phone can designate a port which are free to use 18 Profile Options Use short headers Expose software version Use obsolete transfer mechanism BYE Also Restrict caller identity support varies for proxies from different vendors m Use standard status messages otherwise messages will be taken from SIP packets Voice mail number or address Y Remove fancy characters from phone numbers NENNEN 988998
20. MV 370 MV 372 VoIP GSM Gateway User Manual f MV 379 Mobile Voip COR Tech nati Inc mmunications PORTech Co Content INTRODUCTION tectis ccs hea Ata Seas na e o deg Ana Saag etes Ss cree terae Aa eae aeons a 1 ZUNE TION DESCRIPTION o celeste ep ah Nas Nas UT UE 1 Dob PLR LS ERE NEN 1 4 DIMENSION I4 5CM X 17CM X 3 96 ME aote rite tere tre tessuto tus t ten 2 Da MVe370 PANEL DESCRIPTION sx ssavostephacanstabwostucs ATena tta qi SAID adag TR etsi da tectus Pavo kah ob ana 3 6 MV 3 72 PANEL DESCRIPTION siii id 4 AAN A aah das 5 WE PAGES ETNIA E MUN MENS ee 6 9 SY STEM INFORMATION i eeiscees este ceuen beco ipto eda eh e E PII atin eb ta xen cen aj eoe etu gan 7 TO ROUTE ia is da 7 10 1 MOBILE TO EAN SETTINGS 000 A 8 10 2 CALL BACK SERVICE 50 SETS NEWEBATURE si ees 10 10 3 MOBILE TO LAN SPEED DIAL SETI OSA 1 10 4 LAN TO MOBILE SETTINGS dre tree a a a 12 TSNIOBIEES pasadas uite dte A a etn Cuota e ute 14 LE PNOBIEESTATUS cae quse A cae an haat ee eue ee aa uite 14 11 2 MOBILE SETTINGS iunt treu Uo gt eas A 15 11 3 MOBIEE FORWARD SETTING anta ea pak 17 11 4 MOBILE SMS AGENT zanikajo i dee e ER Pra b a d d po i RR daja a A eva 20 11 5 USE AT COMMAND VIA TELNET OR YOUR PROGRAM eee enne eene et en asse nana nace 21 VENE VOIR AE NE
21. P NMAV370 IP Mask 255 255 255 0 1 Gateway Router IP m DNS Server1 168 95 192 1 DNS Server2 1168 95 1 1 MAC PPPoE Setting User Name Password Submit Reset Here the is important to avoid the two stage dialing when you give a call from Asterisk to GSM LAN To Mobile Table Page 1 Call Num 90 HE ad Y our Asterisk IP OOnNOMWAWN o 45 Mobile To LAN Table Page 1 vi tem Dp URL Select 0 Authorised Mobile 103 1 Another Authorised Mobile 103 2 3 4 5 6 7 8 Z 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 your GSM provider displays the number If you don t know how it is displayed just give a call to the box and check the number given in the Incoming Mob field of the Mobile Status page Any number which is not in that list won t have acces to the LAN side so to Asterisk If you want to allow any number just set in that field but beware of the bill 46 Service Domain Settings Realm 1 Default Active O ON O OFF Display Name 103 m User Name 103 Register Name 1103 Register Password Domain Server Asterisk IP Proxy Server Outbound Proxy Status Not Registered Once Asterisk configuration is made you should get Registered on the Realm1 Codec Settings Codec Priority Codec Priority 1 6 711 ulaw v
22. VITE ACK CANCEL OPTIONS BYE REFER SUBSCRIBE NOTIFY Content Length 0 Transmitting NAT to 192 168 66 203 5060 SIP 2 0 200 OK Via SIP 2 0 UDP 192 168 66 203 5060 branch z9hG4bK672fa67f59c2223275f5ee286d27597a recei ved 192 168 66 203 rport 5060 From lt sip 1002 192 168 66 202 gt tag 4e36d8f1 To lt sip 1002 192 168 66 202 gt tag as13a32ae8 Call ID 7e45b773130f1fc945efcee502f840420192 168 66 203 CSeq 11 REGISTER User Agent Asterisk PBX Allow INVITE ACK CANCEL OPTIONS BYE REFER SUBSCRIBE NOTIFY Expires 300 Contact lt sip 1002 192 168 66 203 5060 gt expires 300 Date Tue 22 May 2007 03 11 54 GMT Content Length 0 59 23 Simple Steps Step 1 Change the Network setting if you need Network network setting Step 2 Register SIP proxy Server or Asterisk or VoipBuster if you need sip setting service domain Step 3 Set Route request mobile to lan 1 gt it is two stage dialing when mobile call in MV 370 MV 372 will provide dial tone and you can enter ip or asterisk extension or phone number If you want to enter phone number please note your asterisk need to have route of destination number 2 specific extension or IP or phone number when mobile call in MV 370 MV 372 will connect with this specific extension or IP or phone number auto If you want to set specific phone number please note your asterisk need to have route of destination number Lan to Mobile 1
23. e Additionally two channels MV 370 MV 372 both need to register proxy server or Asterisk And proxy server asterisk set the route that the prefix of destination number dial out from ch 2 MV 370 MV 372 The channel 2 MV 370 MV 372 s ip the first ip 5062 eg http 192 168 0 100 5062 13 11 Mobile 11 1 Mobile Status PO RTech Mobile Status Your CTI Partner 2008 05 16 18 10 out i era Mobile 1 v Mobile Network Registration Chunghwa Seas SIM Card ID Fwd Settings E SMS Agent Signal Quality 17 Network GSM S N a SIP Settings Incoming IP NAT Transform Incoming IP Name Update Outgoing IP System Authority Save Change Incoming Mob Reboot Outgoing Mob 1 Network Registration The telecom carrier which the SIM card been registered 2 SIM Card ID SIM card ID 3 Signal Quality Signal quality 4 GSM S N IMEI Number Incoming IP The IP address of the last incoming call from LAN 7 Outgoing IP The IP address of the last outgoing call to LAN 8 Incoming Mob The caller ID of the last incoming call from MOBILE 2 3 4 5 6 Incoming IP Name proxy server name 7 8 9 Outgoing Mob The called number of the last outgoing call to MOBILE 14 11 2 Mobile Setting PORTech Your CTI Partner Mobile Setting Route 1 VolP Tx Gain 9 0 12 2 Voip Rx Gain 11 0 15 Mobile 3 LAN Dial
24. ech Communications Inc Save Change Reboot 10 Route Important The route table 50 sets can share by two channels The setting please refer 11 2 Mobile setting ex Mobile 1 use the route table for item 0 24 Mobile 2 use the route table for item 25 49 10 1 Mobile TO LAN Settings The operator may assign 50 sets of routing rule to transfer the call incoming from MOBILE to LAN PORTech mobile To LAN Table Your CTI Partner Route Page v see Mobile To an Speed Dial z O Lan To Mobile Settings Mobile Network SIP Settings NAT Transform Update System Authority Save Change Reboot O O 1000 20UN O Delete Selected Delete All Add New Position D 49 CID Ex 0911111111 0911 URL Ex 192 168 0 1 25t The MV 370 MV 372 will transfer to the URL according to the caller ID of the Mobile CID 1 may enter the whole number e g 0911111111 2 only part of the number prefix e g 0911 means any number starting with 0911 will be accepted 3 means all numbers can be accepted 4 N means the calls without the CID Please note the priority of the rules The item which has more digits will have higher priority If the digits are the same then former one gets the higher priority 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 2 If this field is blank or simply N it means refuse to transfer 3Jf
25. evision reception which can be determined by turning the equipment off and on the user is encouraged to try to correct the interference by one or more of the following measures Reorient or relocate the receiving antenna Increase the separation between the equipment and receiver Connect the equipment into an outlet on a circuit different from that to which the receiver is connected Consult the dealer or an experienced radio TV technician for help Operation is subject to the following two conditions 1 this device may not cause interference and 2 this device must accept any interference including interference that may cause undesired operation of the device 61 FCC RF Radiation Exposure Statement 1 This Transmitter must not be co located or operating in conjunction with any other antenna or transmitter 2 This equipment complies with FCC RF radiation exposure limits set forth for an uncontrolled environment This equipment should be installed and operated with a minimum distance of 20 centimeters between the radiator and your body 62
26. gt It means on progressing When you turn 183 on it means you can hear voicemail while GMS side are busying We recommend you to turn this on if you use SIP Proxy 33 13 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 information When you finished the setting please click the Submit button The QoS setting is to set the voice packets priority If you set the value higher than 0 then the voice packets will get the higher priority to the Internet But the QoS function still need to cooperate with the others Internet devices PORTech Other Settings Your CTI Partner Hold by RFC of Mobile 1 O 0 Off Hold by RFC of Mobile 2 O On Of Route Mobile Network Voice QoS 40 0 63 SIP Settings SIP QoS 40 0 83 Service Domain SIP Expire Time 300 60 86400 sec Port Settings Cadec Settings Codec ID Setting DTMF Setting RPort Setting SIP Responses Other Settings NAT Transform Update System Authority Save Change Reboot 3A 14 NAT Trans In NAT Trans you can setup STUN and uPnP function These functions can help your VoIP device working properly behind NAT 14 1 STUN Setting you can setup the STUN Enable Disable and STUN Server IP address in this page This function can help your VoIP device working properly behind NAT To change these settings please following your ISP information When you fi
27. k up as follows How to stack up 7 2 Connect the antenna and put it in proper position to get the best signal reception 7 3 Insert the SIM card from back of the main body take the slide off first 7 4 Click reset button 3 sec MV 370 MV 372 will restore default IP Other setting as usual 7 5 Connect the power adaptor The POWER LED should be light up 8 Web Page Setting When the IP setting is done the operator may setup all the rest parameters via web page Browse the IP address from Internet Explorer e g http 192 168 0 100 The following page shows up Login PORTech VoIP Enter your username and password to login VoIP server Username Password ES Remember last login Enter the username and password for authentication default username voip password 1234 The page follows when the username and password are correct 9 System Information 9 1 When you login the web page you can see the demo system current system information like firmware version company etc in this page 9 2 Also you can see the function lists in the left side You can use mouse to click the function you want to set up PORTech Mobile VoIP 0 4 Your CTI Partner Route Model Name MV 372 Mobile Model Description GSM 900 1800MHz HO Firmware Version Fri May 16 11 30 35 2008 EN Codec Version Mon Jul 24 10 55 05 2006 SIP Settings NAT Transform Update System Autharity 2007 PORT
28. llowing steps 2 Select the firmware code type Risc code 3 Click the Browse button in the right side of the File Location or you can type the correct path and the filename in File Location blank 4 Select the correct file you want to download to the system then click the Update button 5 Please click update default setting after update firmware PORTech nisi o bert Update Firmware You could update the newest firmware PCB mark 2K123B Route Mobile Method O HTP O TFTP Network SIP Settings NAT Transform Code Type Risc ii Update File Location E Default Settings TFTP Server 192 168 1 250 System Authority Save Change Reboot 38 17 2 Restore Default Settings In this page Update Default Settings you could restore the factory default settings to the system All setting will restore default setting IP will retain original IP as usual not default IP PO RTech Restore Default Settings Your CTI Partner You could click the restore button to restore the factory settings Route eias Restore default settings Network SIP Settings NAT Transform Update New Firmware System Authority Save Change Reboot 39 18 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 will automatically PO RTech Reboot System Your CTI Partner You could press the reboot button to restart the system
29. ne gt show pstn caller id This Is X Lite receiving packet red word is pstn number Test ok INVITE sip 1001 192 168 66 145 7331 SIP 2 0 Via SIP 2 0 UDP 192 168 66 202 5060 branch z9hG4bK3d0bbaf7 rport From 035678238 lt sip 1002 192 168 66 202 gt tag as580472a7 To lt sip 1001 192 168 66 145 7331 gt Contact lt sip 1002 192 168 66 202 gt Call ID 20fa417265e6a26d0b0aae4f551f06f3 9192 168 66 202 CSeq 102 INVITE User Agent Asterisk PBX Max Forwards 70 Date Tue 22 May 2007 02 50 37 GMT Allow INVITE ACK CANCEL OPTIONS BYE REFER SUBSCRIBE NOTIFY 52 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 15852 RTP AVP 0 8 101 a rtpmap 0 PCMU 8000 a rtpmap 8 PCMA 8000 a rtpmap 101 telephone event 8000 a fmtp 101 0 16 a silenceSupp off SIP 2 0 200 Ok Via SIP 2 0 UDP 192 168 66 202 5060 branch z9hG4bK3d0bbaf7 rport From 035678238 lt sip 1002 192 168 66 202 gt tag as580472a7 To lt sip 1001 192 168 66 145 7331 gt tag 677373503 Contact lt sip 1001 192 168 66 145 7331 gt Call ID 20fa417265e6a26d0b0aae4f551f06f3 9192 168 66 202 CSeq 102 INVITE Content Type application sdp Server X Lite release 1105x Content Length 254 v 0 071001 4804366 4807851 IN IP4 192 168 66 145 s X Lite c IN IP4 192 168 66 145 t 0 0 m audio 8000 RTP AVP 0 8 3 101
30. nished the setting please click the Submit button PORTech sTUN Setting Your CTI Partner re STUN of Mobile 1 Oon of STUN of Mobile 2 Oon of Mobile Network STUN Server SIP Settings STUN Port 1024 65535 NAT Transform Update System Authority Save Change Reboot 35 15 System Auth In System Authority you can change your login name and password PORTech Ek CTI vitet System Authority You could change the login usemame password in this page Route New username Mobile New password Network Confirmed password SIP Settings NAT Transform Update Save Change Reboot 36 16 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 Save button After you click the Save button the system will automatically restart and the new setting will effect PO RTech Save Changes Your CTI Partner You have to save changes to effect them Route Mobile Save Changes Network SIP Settings NAT Transform Update New Firmware Default Settings System Authority Save Change Reboot 37 17 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 17 1 Update firmware 1 In New Firmware function you can update new firmware via HTTP in this page You can upgrade the firmware by the fo
31. s 1 CID set the phone number here up to 50 sets 2 URL is the command of call back Application a Call MV 370 MV 372 b MV 370 MV 372 will detect the phone number is in call back list or not c If yes MV 370 MV 372 will reject the call and call it back d You will receive the call from MV 370 MV 372 and prompt a dial tone 10 10 3 Mobile to LAN Speed Dial Settings When you set Mobile to LAN Speed Dial Settings and Mobile to LAN at the same time MV 370 MV 372 will give priority to Mobile to LAN Speed Dial Settings PORTech Mobile To LAN Speed Dial kok cn abad Route A lt A REL 0 Test 192 168 0 107 Mobile To Lan Setting 1 Mobile To Lan Speed Dial an To Mobile Settings 2 Mobile 3 4 Network 5 SIP Settings B NAT Transform i Update 8 System Authority 3 E Reboot 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 Num 0 Name test URL 192 168 0 107 When the caller hear dial tone and enter 0 system will connect 192 168 0 107 11 10 4 LAN to Mobile Settings The operator may assign 50 sets of routing rule to transfer the call incoming from LAN to MOBILE PORTech AN To Mobile Table Your CTI Partner Route Page 1 Y eae eee ME E CNN Call Num O Mobile To Lan Speed Dial Lan To Mobile Settings Mobile SIP Settings NAT Transform Update System Authority
32. s Codec Priority 5 G 726 16 wj Codec Settings Codec Priority 8 G726 24 Codec ID Setting DTMF Setting Codec Priority 7 5 726 32 hl RPort Setting Codec Priority 8 G 726 40 SIP Responses Other Settings RTP Packet Length NAT Transform 6 711 amp 0 9 20 ms v Update G 723 30 ms v System Authority Save Change G 723 5 3K G 723 5 3K On of Reboot Voice VAD Voice VAD OOn Off 29 13 4 Codec ID Setting You can setup the Codec ID in this page PORTech Codec ID Setting Your CTI Partner You could set the value of Codec ID in this page Route Mobile Codec Type o WDefault Value Network 6726 16 ID 23 95 255 23 SIP Settings 6726 24 ID 22 95 255 2 6726 32 ID 2 2 Service Domain M a Port Settings 6726 40 ID 21 85 265 21 oe RFC 2893 ID 10 95 255 101 m DTMF Setting RPort Setting SIP Responses Other Settings NAT Transform Update System Authority Save Change Reboot 30 13 5 DTMF Setting You can setup the DTMF Setting in this page DTMF Setting Mobile DTMF Transfer to Lan 9 2833 O Inband DTMF O Send DTMF SIP Info 31 PORTech Your CTI Partner Route Mobile Network SIP Settings Service Domain Port Settings Codec Settings Codec ID Setting DTMF Setting RPort Setting SIP Responses Other Settings NAT Transform Update 8
33. secret 1001 qualify yes nat yes host dynamic canreinvite no context internal 1002 type friend 50 secret 1002 qualify yes nat yes host dynamic canreinvite no context internal e Add the following setting to etc asterisk extensions conf internal exten gt 1000 1 Dial SIP 1000 exten gt 1001 1 Dial SIP 1001 exten gt 1002 1 Dial SIP 1002 configure trixbox 2 2 address 192 168 66 202 5060 SJPhone address 192 168 66 145 5060 username 1000 displayname user_1000 X Lite address 192 168 66 145 7331 username 1001 displayname user_1001 MV 370 MV 372 address 192 168 66 203 5060 username 1002 displayname user_1002 5 mla 5 Search Trance GPR a le Ca NIE 8 3 9 Realm 1 Default Act O arar 100 in this page Explorer Service Domain Settings of service domains On C Of juser 1002 1002 1002 192 168 66 202 192 168 66 202 192 168 66 202 egistered You could set information No Mobile 1 y Windows Internet Exp 92 168 66 203 ogin e Management VoIP Web Management Windo Co D E nt Jn BRO HE KAO REG TAO HAD ve E Yo Web Mobile Voip x Route Mobile Network SIP Settings NAT Trans System Auth Save Change Update Reboot E test1 pstn gt call 092849291 1 mobile number gt MV 370 MV 372 gt hear the second dial tone call SoftPhone s number gt SoftPho
34. t the Username and the Password correctly 3 The Bridge Item is to setuo the system Bridge mode Enable Disable If you set the Bridge On then the two Fast Ethernet ports will be transparent 4 When you finished the setting please click the Submit button 22 PO RTech WAN Settings Your CTI Partner You could configure the WAN settings in this page Route Network Mode O Bridge 9 NAT Mobile Rahman WAN Setting AA IP Type O Fixed IP O DHCP Client O PPPoE atus WAN Settings IP 192 168 0 122 LAN Settings Mask 255 255 255 0 SNTP Settings Gateway 192 168 0 254 6 DNS Server 168 95 192 1 NAT Transform DNS Server 168 95 1 1 Update MAC 00037 e009999 System Authority Save Change PPPoE Setting Reboot User Name Password 12 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 network environment You may refer to your current network environment to configure the system properly 2 DHCP Server You may refer to your current network environment to configure the system properly 23 LAN Settings IP 192 168 0 102 1 Mask 255 255 255 0 MAC 00037 e008888 DHCP Server DHCP Server OOn Gor Start IP 150 End IP 200 Lease Time 1 0 dd hh Submit Reset 24 PORTech Your CTI Partner Route Mobile Network Status VVAN Settings
35. the tree SIP account First you need to click Active to enable the Service Domain then you can input the following items 1 No choose Mobile 1 or Mobile 2 2 Display name you can input the name you want to display 3 User name you need to input the User Name get from your ISP 4 Register Name you need to input the Register Name get from your ISP 5 Register Password you need to input the Register Password get from ISP 6 Domain Server you need to input the Domain Server get from your ISP 7 Proxy Server you need to input the Proxy Server get from your ISP 8 Outbound Proxy you need to input the Outbound Proxy get from your ISP If your ISP does not provide the information then you can skip this item 9 You can see the Register Status in the Status item 10 When you finished the setting please click the Submit button Remember to click Save Charge 26 PORTech MEE Service Domain Settings Mobile 1 v Route ee Network Active O ON O OFF SIP Settings Display Name 3001 ervice Domain Port Settings Register Narne 3001 EE tte Register Password esco Codec ID Setting DTMF Setting Domain Server 1 RPort Setting Proxy Server 61 218 151 230 SIP Responses Other Settings Outbound Proxy NAT Transform Status Not Registered Example Register VoipBuster Realm 1 Default Active On Off Display Name fenyo User Name fienny0922 Your Voipb
36. tone Gain 9 0 12 Status Settings 4 Mobile 0 1 NON Fwd Settings 5 Routing Range 0 to 43 0 49 SMS Agent I 0 z 6 CODEC Tx Gain 6 0 7 7 CODEC Rx Gain 6 0 7 Network 8 SIP From Tel User Standard v Answer Delay 0 0 15 12 SIP Settings 9 CLID Presentation Suppression Invocation NAT Transform 10 Mobile PIN Code On E Code Confirmed Update LAN Answer Mode Answered Alerted Income 11 Routing Range 0 to 49 D A9 CODEC Tx Gain CODEC Rx Gain 6 0 7 6 e SIP From Tel User Standard Answer Delay D 0 15 CLID Presentation O Suppression Invocation Mobile PIN Code On Code Confirmed LAN Answer Mode Answered Alerted Income Mobile 1 _ System Authority Save Change Reboot 1 VoIP Tx Gain 4 2 VoIP Rx Gain 1 VoIP Tx Gain To adjust the volume of LAN side 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 If you use this channel please click on Otherwise please click off 5 Routing Range The route table 50 sets can share by two channels ex Mobile 1 use the route table for item 0 24 Mobile 2 use the route table for item 25 49 6 CODEC Tx Gain as above 7 CODEC Rx Gain as above 8 SIP From Caller ID transfer e Tel User Standard
37. uster username Register Name 60010922 Register Password Your Voipbuster password Domain Server NANJ Proxy Server 194 221 62 207 Proxy Server s IP _ Outbound Proxy FO Status Registered 27 13 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 refer to the ISP to setup the port number correctly When you finished the setting please click the Submit button PORTech Ports Setting Your CTI Partner Route Port of Mobile 1 Mobile SIP Port 5060 1024 65535 Network RTP Port 60000 1024 55535 rends Service Domain SIP Port 15062 1024 65535 Port Settings eoo E RTP Port 60100 1024 65535 UJE E 5 Codec ID Setting DTMF Setting RPort Setting SIP Responses Other Settings NAT Transform Update System Authority Save Change Reboot 28 13 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 items When you finished the setting please click the Submit button PO RTech Your CTI Partner Codec Settings Route Codec Priority Mobile Codec Priority 1 6 711 u law v Network Codec Priority 2 16 711 a law v SIP Settings Codec Priority 3 AES ul Codec Priority 4 G 728 v Service Domain Port Setting
38. version of the firmware running The system will change to DHCP Client type DHCP will be disabled and system will change to the Static IP type Enter IP address using numbers on the telephone key pad Use the star key when entering a decimal point Must set Static IP first Enter value using numbers on the telephone key pad Use the star key when entering a decimal point Must set Static IP first Enter IP address using numbers on the telephone key pad Use the star key when entering a decimal point Must set Static IP first Enter IP address using numbers on the telephone key pad Use the star key when entering a decimal point 111284 521112 41120000 xxx 411 3000 xxx 41140000 xxx 1 15xxx XXX XXX oodt 42 DNS Server Check Firmware Version Set as client DHCP Set Static IP Address Set Network Mask Set Gateway IP Address Set Primary DNS Server 10 11 12 13 20 Specification 20 1 Protocols SIP RFC2543 RFC3261 20 2 TCP IP IP TCP UDP RTP RTCP CMP ARP RARP SNTP DHCP DNS Client IEEE802 1P Q ToS DiffServ NAT Traversal STUN uPnP IP Assignment Static IP DHCP PPPoE 20 3 Codec G 711 u Law G 711 a Law G 723 1 5 3k G 723 1 6 3k G 729A G 729A B 20 4 Voice Quality VAD 43 CNG AEC LEC Packet loss 20 5 GSM MV 370 MV 372 Dual BAND 900 1800 MHZ Tri BAND BenQ M23 900 1800 19
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