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Add the rest of the Grandstream provisioning files.

This commit is contained in:
Mark Crane 2013-12-07 03:06:02 +00:00
parent a643c72b62
commit 3fc22fe724
25 changed files with 54957 additions and 1064 deletions
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<?xml version="1.0" encoding="utf-8"?>
<gs_provision version="1">
<config version="1">
<!-- Configuration template for GXV-300x firmware version 1.2.3.7 -->
<!-- Advanced/System-wide Options -->
<!-- Admin password for web interface, max. length is 30 -->
<P2>admin</P2>
<!-- G.723 rate: 0 - 6.3kbps encoding rate, 1 - 5.3kbps encoding rate -->
<P49></P49>
<!-- Silence Suppression. 0 - no, 1 - yes -->
<P50>0</P50>
<!-- Voice Frames per TX (up to 10/20/32/64 frames for G711/G726/G723/other codecs respectively) -->
<P37>2</P37>
<!-- Video frame rate: default - 15 -->
<P904>15</P904>
<!-- Video bit rate (kbps): default is 128 -->
<P905>128</P905>
<!-- Video Packet Size: maxlength = 4, from 100~1400, default is 1400 -->
<P927>800</P927>
<!-- Video Rate Control. 0 - Frame, 1 - TMN8, 2 - GOP (default is 0) -->
<P924>0</P924>
<!-- Video Frame Skipping. 0 - no, 1 - yes -->
<P925>0</P925>
<!-- Sharpening Filter. (0-9, 0 means disabled, default is 5) -->
<P938>5</P938>
<!-- Brightening Filter. (0-9, 0 means disabled, default is 5) -->
<P939>5</P939>
<!-- Tone Remapping Filter. (0-255, 0 means disabled, default is 32) -->
<P948>32</P948>
<!-- Packetization-mode. 0 - 1, 1 - 1 -->
<P957>0</P957>
<!-- Enable Video Surveillance. Turn phone into video surveillance camera if set to 1, YES. Default=No, 0. -->
<P928>0</P928>
<!-- Surveillance RTSP Port. Video Surveillance data stream port, default is 554. -->
<P929>554</P929>
<!-- Streaming RTSP Server -->
<P953></P953>
<!-- Streaming RTSP User ID -->
<P954></P954>
<!-- Streaming RTSP Password -->
<P955></P955>
<!-- Layer 3 QoS (IP Diff-Serv or Precedence value for RTP) -->
<P38>0</P38>
<!-- Layer 2 QoS. 802.1Q/VLAN Tag (VLAN classification for RTP) -->
<P51>0</P51>
<!-- Layer 2 QoS. 802.1p priority value (0 - 7) -->
<P87>0</P87>
<!-- No Key Entry Timeout. Default - 4 seconds. -->
<P85>4</P85>
<!-- Use # as Dial Key. 0 - no, 1 - yes -->
<P72>1</P72>
<!-- Local RTP port (1024-65535, default 5004) -->
<P39>5004</P39>
<!-- Use Random Port. 0 - no, 1 - yes -->
<P78>0</P78>
<!-- Keep-alive interval (in seconds. default 20 seconds) -->
<P84>20</P84>
<!-- Use NAT IP. This will enable our SIP client to use this IP in the SIP message. Example 64.3.153.50. -->
<P101></P101>
<!-- STUN server -->
<P76></P76>
<!-- Firmware Upgrade -->
<!-- Firmware Upgrade. 0 - TFTP Upgrade, 1 - HTTP Upgrade. -->
<P212>1</P212>
<!-- Firmware Server Path -->
<P192>fm.grandstream.com/gs</P192>
<!-- Config Server Path -->
<P237>{$domain_name}{$project_path}/app/provision</P237>
<!-- Firmware File Prefix -->
<P232></P232>
<!-- Firmware File Postfix -->
<P233></P233>
<!-- Config File Prefix -->
<P234></P234>
<!-- Config File Postfix -->
<P235></P235>
<!-- Allow DHCP Option 66 to override server. 0 - No, 1 - Yes. Default is Yes. -->
<!-- When set to Yes(1), it will override the configured provision path and method. -->
<P145>1</P145>
<!-- Automatic Upgrade. 0 - No, 1 - Yes. Default is No. -->
<P194>0</P194>
<!-- Check for new firmware every () minutes, unit is in minute, minimnu 60 minutes, default is 7 days. -->
<P193>10080</P193>
<!-- Use firmware pre/postfix to determine if f/w is required -->
<!-- 0 = Always Check for New Firmware -->
<!-- 1 = Check New Firmware only when F/W pre/suffix changes -->
<!-- 2 = Always Skip the Firmware Check -->
<P238>0</P238>
<!-- Enable Screen Saver Download -->
<!-- 0 = No -->
<!-- 1 = YES, HTTP -->
<!-- 2 = YES, TFTP -->
<P933>1</P933>
<!-- Screen Saver Download Server Path -->
<P934>fm.grandstream.com/gs</P934>
<!-- Enable Phonebook XML Download -->
<!-- 0 = No -->
<!-- 1 = YES, HTTP -->
<!-- 2 = YES, TFTP -->
<P330>0</P330>
<!-- Phonebook XML Server Path -->
<!-- This is a string of up to 128 characters that should contain a path to the XML file -->
<!-- It MUST be in the host/path format. For example: "directory.grandstream.com/engineering" -->
<P331></P331>
<!-- Remove Manually-edited entries on Download -->
<!-- 0 - No, 1 - Yes -->
<P333>0</P333>
<!-- Enable Network Screen Saver -->
<!-- 0 - No -->
<!-- 1 - YES, HTTP -->
<P943>0</P943>
<!-- Network Screen Saver Server Path -->
<P942></P942>
<!-- Enable Weather Forecast -->
<!-- 0 - No -->
<!-- 1 - YES, HTTP -->
<P945>0</P945>
<!-- Weather Forecast Server Path -->
<P944></P944>
<!-- Enable Headline News -->
<!-- 0 - No -->
<!-- 1 - YES, HTTP -->
<P947>0</P947>
<!-- Headline News RSS Server Path -->
<P946></P946>
<!-- Offhook Auto Dial -->
<P71></P71>
<!-- DTMF Payload Type -->
<P79>101</P79>
<!-- Syslog Server (name of the server, max length is 64 charactors) -->
<P207></P207>
<!-- Syslog Level (Default setting is NONE) -->
<!-- 0 - NONE, 1 - DEBUG, 2 - INFO, 3 - WARNING, 4 - ERROR -->
<P208>0</P208>
<!-- NTP Server -->
<P30>us.pool.ntp.org</P30>
<!-- Allow DHCP Option 42 to override NTP server. 0 - No, 1 - Yes. Default is No. -->
<!-- When set to Yes(1), it will override the configured NTP server. -->
<P144>0</P144>
<!-- Distinctive Ring Tone -->
<!-- Use custom ring tone 1 if incoming caller ID is the following: -->
<P105></P105>
<!-- Use custom ring tone 2 if incoming caller ID is the following: -->
<P106></P106>
<!-- Use custom ring tone 3 if incoming caller ID is the following: -->
<P107></P107>
<!-- System Ring Tone: input type = text, maxlength=64 size=30 -->
<P345>f1=440,f2=480,c=200/400;</P345>
<!--Call Progress Tones: input type=text, maxlength=64, size=30 -->
<!-- Dial Tone -->
<P343>f1=350,f2=440;</P343>
<!-- Message Waiting Tone -->
<P344>f1=350,f2=440,c=10/10;</P344>
<!-- Ring Back Tone -->
<P346>f1=440,f2=480,c=200/400;</P346>
<!-- Call-Waiting Tone -->
<P347>f1=440,f2=440,c=25/525;</P347>
<!-- Busy Tone -->
<P348>f1=480,f2=620,c=50/50;</P348>
<!-- Reorder Tone -->
<P349>f1=480,f2=620,c=25/25;</P349>
<!-- Start Browser On Boot: 0 - no (default), 1 - yes -->
<P950>0</P950>
<!-- Enable Browser Toolbar: 0 - no (default), 1 - yes -->
<P951>1</P951>
<!-- Browser Home Page, maxlength=64, default = about:/start.htm -->
<P952></P952>
<!-- Disable Call Waiting. 0 - no, 1 - yes -->
<P91>0</P91>
<!-- Disable Call Waiting Tone: 0 - no (default), 1 - yes -->
<P186>0</P186>
<!-- Disable Direct IP Call. 0 - no, 1 - yes -->
<P277>0</P277>
<!-- Use Quick IP-call mode: 0 - no, 1 - yes. Default is No. -->
<P184>0</P184>
<!-- Allow Media Loopback. 0 - no, 1 - yes -->
<P278>0</P278>
<!-- Lock Keypad Update. 0 - no, 1 - yes -->
<P88>0</P88>
<!-- Display Language. 0 - English, 1 - Downloaded Language, 2 - Chinese -->
<P342>0</P342>
<!-- language file postfix -->
<P399></P399>
<!-- Primary Account (Account 1) Settings -->
<!-- Account Active (In Use). 0 - no, 1 - yes -->
<P271>1</P271>
<!-- Account Name -->
<P270></P270>
<!-- SIP Server: max. length = 96 -->
<P47>{$server_address_1}</P47>
<!-- Outbound Proxy -->
<P48>proxy.mycompany.com</P48>
<!-- SIP User ID -->
<P35>{$user_id_1}</P35>
<!-- Authenticate ID -->
<P36>{$user_id_1}</P36>
<!-- Authenticate password -->
<P34>{$user_password_1}</P34>
<!-- Display Name (John Doe) -->
<P3>{$display_name_1}</P3>
<!-- Use DNS SRV. 0 - No, 1 - Yes. -->
<P103>0</P103>
<!-- SIP User ID is phone number. 0 - no, 1 - yes -->
<P63>0</P63>
<!-- SIP Registration. 0 - no, 1 - yes -->
<P31>1</P31>
<!-- Unregister On Reboot. 0 - no, 1 - yes -->
<P81>0</P81>
<!-- Register Expiration (in minutes. default 1 hour, max 45 days) -->
<P32>3</P32>
<!-- SIP Registration Failure Retry Wait Time (in seconds. Between 1-3600, default is 20) -->
<P138>20</P138>
<!-- Local SIP port (default 5060) -->
<!-- SIP Transport 1 - UDP(default), 2 - TCP -->
<P130>1</P130>
<!-- NAT Traversal (STUN). 0 - yes, 1 - no, 2 - No, but send keep-alive -->
<P52>1</P52>
<!-- Keep-Alive Using SIP OPTIONS. 0 - no, 1 - yes -->
<P1309>0</P1309>
<!-- SUBSCRIBE for MWI. (Whether or not send SUBSCRIBE for Message Waiting Indication) 0 - No, 1 - Yes. -->
<P99>0</P99>
<!-- PUBLISH for Presence, 0 - no (default), 1 - yes -->
<P188>0</P188>
<!-- Proxy-Require (A SIP extension to enable firewall penetration) -->
<P197></P197>
<!-- SIP Compact Header. 0 - no, 1 - yes -->
<P289>0</P289>
<!-- Voice Mail UserID (User ID/extension for 3rd party voice mail system) -->
<P33></P33>
<!-- Send DTMF. 8 - in audio, 1 - via RTP, 2 - via SIP INFO, 11 - In Audio & RTP & SIP INFO -->
<!-- 9 - In Audio & RTP, 10 - IN Audio & SIP INFO, 3 - RTP & SIP INFO -->
<P73>8</P73>
<!-- Early Dial (use "Yes" only if proxy supports 484 response). 0 - no, 1 - yes -->
<P29>0</P29>
<!-- Dial Plan Prefix (dial plan prefix string) -->
<P66></P66>
<!-- Dial Plan. Maxlength = 128 character. Defalut is allow all {x+|*x+} -->
<P290>{x+|*x+}</P290>
<!-- Delayed Call Forward Wait Time. Default is 20 seconds, maxlength = 5 -->
<P139>20</P139>
<!-- Enable Call Features. 0 - no, 1 - yes -->
<P191>1</P191>
<!-- Session Expiration (in seconds. default 180 seconds. Allowed value: 90-65535) -->
<P260>180</P260>
<!-- Minimum SE (in seconds. default 90 seconds, must be lower than or equal to P260) -->
<P261>90</P261>
<!-- Caller Request Timer (Request for timer when calling) 0 - no, 1 - yes -->
<P262>0</P262>
<!-- Callee Request Timer (Request for timer when called. i.e. if remote party supports timer but did not request for one) 0 - no, 1 - yes -->
<P263>0</P263>
<!-- Force Timer (Still use timer when remote party does not support timer) 0 - no, 1 - yes -->
<P264>0</P264>
<!-- UAC Specify Refresher. 0 - omit, 1 - UAC, 2 - UAS -->
<P266>0</P266>
<!-- UAS Specify Refresher. 1 - UAC, 2 - UAS -->
<P267>1</P267>
<!-- Force INVITE (Always refresh with INVITE instead of UPDATE even when remote party supports UPDATE) 0 - no, 1 - yes -->
<P265>0</P265>
<!-- Enable 100rel. 0 - no, 1 - yes -->
<P272>0</P272>
<!-- Audio Call Ring Tone. 0 - system ring tone, 1 - custom ring tone 1, 2 - custom ring tone 2, 3 - custom ring tone 3. -->
<P104>0</P104>
<!-- Video Call Ring Tone. 0 - system ring tone, 1 - custom ring tone 1, 2 - custom ring tone 2, 3 - custom ring tone 3. -->
<P1354>0</P1354>
<!-- Send Anonymous. 0 - no, 1 - yes (caller ID will be blocked if set to Yes) -->
<P65>0</P65>
<!-- Anonymous Call Rejection: 0 - no (default), 1 - yes -->
<P129>0</P129>
<!-- Check SIP User ID for incoming INVITE, 0 - no (default), 1 - yes -->
<P258>0</P258>
<!-- Refer-To Uses Target Contact: 0 - no (default), 1 - yes -->
<P135>0</P135>
<!-- Disable Multiple Media Attribute in SDP, 0 - no (default), 1 - yes -->
<P137>0</P137>
<!-- Auto Answer. 0 - no, 1 - yes, 2 - Intercom/Paging -->
<P90>0</P90>
<!-- Preferred Vocoder -->
<!-- 0 - PCMU, 2 - G.726-32, 3 - GSM, 4 - G.723.1, 8 - PCMA, -->
<!-- 9 - G.722, 18 - G.729A/B -->
<!-- Currently only PCMU/PCMA, GSM, G.723, G.729 and G.726-32 supported, will add others in the future -->
<!-- First codec. -->
<P57>0</P57>
<!-- Second codec. -->
<P58>8</P58>
<!-- Third codec. -->
<P59>4</P59>
<!-- Forth codec. -->
<P60>18</P60>
<!-- Fifth codec. -->
<P61>3</P61>
<!-- Sixth codec. -->
<P62>2</P62>
<!-- Seventh codec. -->
<P46>0</P46>
<!-- Eighth codec. -->
<P98>0</P98>
<!-- Preferred Video Coder: 99 - H.264, 34 - H.263, 103 - H.263+ (1998) -->
<!-- Choice 1 -->
<P295>99</P295>
<!-- Choice 2 -->
<P296>34</P296>
<!-- Choice 3 -->
<P1307>103</P1307>
<!-- Choose Video Codec By Local Preference. 0 - No, 1 - Yes -->
<P1308>0</P1308>
<!-- Jitter Delay: 0 - Medium, 1 - Low, 2 - High -->
<P300>0</P300>
<!-- Enable Video: 0 - No, 1 - Yes, 2 - No, but allow in-call enabling -->
<P292>1</P292>
<!-- H.264 payload type: between 96 ~ 127, default = 99 -->
<P293>99</P293>
<!-- H.263+ (1998) payload type: between 96 ~ 127, default = 103 -->
<P350>103</P350>
<!-- H.263 Default Resolution. 0 - CIF, 1 - QCIF -->
<P1330>0</P1330>
<!-- Enable RFC5168 Support. 0 - No, 1 - Yes -->
<P1331>0</P1331>
<!-- SRTP Mode: Disabled - 0, Enabled but not forced - 1, Enabled and forced - 2 -->
<P183>0</P183>
<!-- Special Feature. 100 - Standard, 101 - Nortel MCS, 102 - BroadSoft, 104 - Sonus, 108 - CBCOM, 109 - RNK -->
<P198>100</P198>
<!-- SIP Account 2 Settings -->
<!-- Account Active (In Use). 0 - no, 1 - yes -->
<P401>0</P401>
<!--Account Name -->
<P417>0</P417>
<!-- SIP Server -->
<P402>{$server_address_2}</P402>
<!-- Outbound Proxy Server -->
<P403></P403>
<!-- Account 2 SIP User ID -->
<P404>{$user_id_2}</P404>
<!-- Authenticate ID -->
<P405>{$user_id_2}</P405>
<!-- Authenticate Password -->
<P406>{$user_password_2}</P406>
<!-- Display Name (John Doe) -->
<P407>{$display_name_2}</P407>
<!-- Use DNS SRV. 0 - No, 1 - Yes -->
<P408>0</P408>
<!-- User ID is phone number; "user=phone". 0 - no, 1 - yes -->
<P409></P409>
<!-- SIP Registration. 0 - no, 1 - yes -->
<P410>1</P410>
<!-- Unregister on Reboot. 0 - no, 1 - yes -->
<P411>0</P411>
<!-- Register Expiration (in minutes. default 1 hour, max 45 days) -->
<P412>3</P412>
<!-- Registration Retry Wait Time (in seconds. Between 1-3600, default is 20) -->
<P471>20</P471>
<!-- SIP Transport, 1 - UDP(default), 2 - TCP -->
<P448>1</P448>
<!-- Local SIP port (default 5062) -->
<P413>5062</P413>
<!-- NAT Traversal. 0 - yes, 1 - no, 2 - No, but send keep-alive -->
<P414>1</P414>
<!-- Keep-Alive Using SIP OPTIONS. 0 - no, 1 - yes -->
<P490>0</P490>
<!-- Subscribe MWI. (Whether or not send SUBSCRIBE for Message Waiting Indication) 0 - No, 1 - Yes. -->
<P415>0</P415>
<!-- PUBLISH for Presence, 0 - no (default), 1 - yes -->
<P488>0</P488>
<!-- Proxy Require -->
<P418></P418>
<!-- SIP Compact Header. 0 - no, 1 - yes -->
<P472>0</P472>
<!-- Voice Mail UserID (User ID/extension for 3rd party voice mail system) -->
<P426></P426>
<!-- Send DTMF. 8 - in audio, 1 - via RTP, 2 - via SIP INFO, 11 - In Audio & RTP & SIP INFO -->
<!-- 9 - In Audio & RTP, 10 - IN Audio & SIP INFO, 3 - RTP & SIP INFO -->
<P416>8</P416>
<!-- Early Dial. 0 - no, 1 - yes -->
<P422>0</P422>
<!-- Dial Plan Prefix -->
<P419></P419>
<!-- Dial Plan. Maxlength = 128, default is allow all {x+|*x+}-->
<P459>{x+|*x+}</P459>
<!-- Delayed Call Forward Wait Time. Default is 20 seconds, maxlength = 5 -->
<P470>20</P470>
<!-- Enable Call Features. 0 - no, 1 - yes.-->
<P420>1</P420>
<!-- Session Expiration (in seconds. default 180 seconds. Allowed value: 90-65535) -->
<P434>180</P434>
<!-- Minimum SE (in seconds. default 90 seconds, must be lower than or equal to P260) -->
<P427>90</P427>
<!-- Caller Request Timer (Request for timer when calling) 0 - no, 1 - yes -->
<P428>0</P428>
<!-- Callee Request Timer (Request for timer when called. i.e. if remote party supports timer but did not request for one) 0 - no, 1 - yes -->
<P429>0</P429>
<!-- Force Timer (Still use timer when remote party does not support timer) 0 - no, 1 - yes -->
<P430>0</P430>
<!-- UAC Specify Refresher. 0 - omit, 1 - UAC, 2 - UAS -->
<P432>0</P432>
<!-- UAS Specify Refresher. 1 - UAC, 2 - UAS -->
<P433>1</P433>
<!-- Force INVITE (Always refresh with INVITE instead of UPDATE even when remote party supports UPDATE) 0 - no, 1 - yes -->
<P431>0</P431>
<!-- Enable 100rel. 0 - no, 1 - yes -->
<P435>0</P435>
<!-- Audio Call Ring Tone. 0 - system ring tone, 1 - custom ring tone 1, 2 - custom ring tone 2, 3 - custom ring tone 3. -->
<P423>0</P423>
<!-- Video Call Ring Tone. 0 - system ring tone, 1 - custom ring tone 1, 2 - custom ring tone 2, 3 - custom ring tone 3. -->
<P482>0</P482>
<!-- Send Anonymous. 0 - no, 1 - yes ((caller ID will be blocked if set to Yes) -->
<P421>0</P421>
<!-- Anonymous Call Rejection: 0 - no (default), 1 - yes -->
<P446>0</P446>
<!-- Check SIP User ID for incoming INVITE, 0 - no (default), 1 - yes -->
<P449>0</P449>
<!-- Refer-To Uses Target Contact: 0 - no (default), 1 - yes -->
<P469>0</P469>
<!-- Disable Multiple Media Attribute in SDP, 0 - no (default), 1 - yes -->
<P487>0</P487>
<!-- Auto Answer. 0 - no, 1 - yes, 2 - Intercom/Paging -->
<P425>0</P425>
<!-- Preferred Vocoder -->
<!-- 0 - PCMU, 2 - G.726-32, 3 - GSM, 4 - G.723.1, 8 - PCMA, -->
<!-- 9 - G.722, 18 - G.729A/B -->
<!-- Currently only PCMU/PCMA, GSM, G.723, G.729 and G.726-32 supported, will add others in the future -->
<!-- First codec. -->
<P451>0</P451>
<!-- Second codec. -->
<P452>8</P452>
<!-- Third codec -->
<P453>4</P453>
<!-- Forth codec.-->
<P454>18</P454>
<!-- Fifth codec.-->
<P455>3</P455>
<!-- Sixth codec. -->
<P456>2</P456>
<!-- Seventh codec. -->
<P457>0</P457>
<!-- Eighth codec. -->
<P458>0</P458>
<!-- Preferred Video Codec: 99 - H.264, 34 - H.263 103 - H.263+ (1998) -->
<!-- Choice 1 -->
<P464>99</P464>
<!-- Choice 2 -->
<P465>34</P465>
<!-- Choice 3 -->
<P475>103</P475>
<!-- Choose Video Codec By Local Preference. 0 - No, 1 - Yes -->
<P474>0</P474>
<!-- Jitter Delay: 0 - Medium, 1 - Low, 2 - High -->
<P467>0</P467>
<!-- Enable Video: 0 - No, 1 - Yes, 2 - No, but allow in-call enabling -->
<P461>1</P461>
<!-- H.264 payload type: between 96 ~ 127, default = 99 -->
<P462>99</P462>
<!-- H.263+ payload type: between 96 ~ 127, default = 103 -->
<P473>103</P473>
<!-- H.263 Default Resolution. 0 - CIF, 1 - QCIF -->
<P477>0</P477>
<!-- Enable RFC5168 Support. 0 - No, 1 - Yes -->
<P478>0</P478>
<!-- SRTP Mode: Disabled - 0, Enabled but not forced - 1, Enabled and forced - 2 -->
<P443>0</P443>
<!-- Special Feature. 100 - Standard, 101 - Nortel MCS, 102 - BroadSoft, 104 - Sonus, 108 - CBCOM, 109 - RNK -->
<P424>100</P424>
<!-- SIP Account 3 Settings -->
<!-- Account Active (In Use). 0 - no, 1 - yes -->
<P501>0</P501>
<!-- Account Name -->
<P517></P517>
<!-- SIP Server -->
<P502>{$server_address_3}</P502>
<!-- Outbound Proxy Server -->
<P503></P503>
<!-- SIP User ID -->
<P504>{$user_id_3}</P504>
<!-- Authenticate ID -->
<P505>{$user_id_3}</P505>
<!-- Authenticate Password -->
<P506>{$user_password_3}</P506>
<!-- Display Name (John Doe) -->
<P507>{$display_name_3}</P507>
<!-- Use DNS SRV. 0 - No, 1 - Yes. -->
<P508>0</P508>
<!-- User ID is phone. "user=phone". 0 - no, 1 - yes -->
<P509>0</P509>
<!-- SIP Registration. 0 - no, 1 - yes -->
<P510>1</P510>
<!-- Unregister on Reboot. 0 - no, 1 - yes -->
<P511>0</P511>
<!-- Register Expiration (in minutes. default 1 hour, max 45 days) -->
<P512>3</P512>
<!-- Registration Retry Wait Time (in seconds. Between 1-3600, default is 20) -->
<P571>20</P571>
<!-- SIP Local port (default 5064) -->
<P513>5064</P513>
<!-- SIP Transport, 1 - UDP(default), 2 - TCP -->
<P548>1</P548>
<!-- NAT Traversal. 0 - yes, 1 - no, 2 - No, but send keep-alive -->
<P514>1</P514>
<!-- Keep-Alive Using SIP OPTIONS. 0 - no, 1 - yes -->
<P590>0</P590>
<!-- Subscribe MWI. (Whether or not send SUBSCRIBE for Message Waiting Indication) 0 - no, 1 - yes -->
<P515>0</P515>
<!-- PUBLISH for Presence, 0 - no (default), 1 - yes -->
<P588>0</P588>
<!-- Proxy Require -->
<P518></P518>
<!-- SIP Compact Header. 0 - no, 1 - yes -->
<P572>0</P572>
<!-- Voice Mail UserID (User ID/extension for 3rd party voice mail system) -->
<P526></P526>
<!-- Send DTMF. 8 - in audio, 1 - via RTP, 2 - via SIP INFO, 11 - In Audio & RTP & SIP INFO -->
<!-- 9 - In Audio & RTP, 10 - IN Audio & SIP INFO, 3 - RTP & SIP INFO -->
<P516>8</P516>
<!-- Early Dial. 0 - no, 1 - yes -->
<P522>0</P522>
<!-- Dial Plan Prefix -->
<P519></P519>
<!-- Dial Plan: Maxlength = 128, default is allow all {x+|*x+} -->
<P559>{x+|*x+}</P559>
<!-- Delayed Call Forward Wait Time. Default is 20 seconds, maxlength = 5 -->
<P570>20</P570>
<!-- Enable Call Features. 0 - no, 1 - yes. -->
<P520>1</P520>
<!-- Session Expiration (in seconds. default 180 seconds. Allowed value: 90-65535) -->
<P534>180</P534>
<!-- Minimum SE (in seconds. default 90 seconds, must be lower than or equal to P260) -->
<P527>90</P527>
<!-- Caller Request Timer (Request for timer when calling) 0 - no, 1 - yes -->
<P528></P528>
<!-- Callee Request Timer (Request for timer when called. i.e. if remote party supports timer but did not request for one) 0 - no, 1 - yes -->
<P529></P529>
<!-- Force Timer (Still use timer when remote party does not support timer) 0 - no, 1 - yes -->
<P530></P530>
<!-- UAC Specify Refresher. 0 - omit, 1 - UAC, 2 - UAS -->
<P532>0</P532>
<!-- UAS Specify Refresher. 1 - UAC, 2 - UAS -->
<P533>1</P533>
<!-- Force INVITE (Always refresh with INVITE instead of UPDATE even when remote party supports UPDATE) 0 - no, 1 - yes -->
<P531></P531>
<!-- Enable 100rel. 0 - no, 1 - yes -->
<P535>0</P535>
<!-- Audio Call Ring Tone. 0 - system ring tone, 1 - custom ring tone 1, 2 - custom ring tone 2, 3 - custom ring tone 3. -->
<P523>0</P523>
<!-- Video Call Ring Tone. 0 - system ring tone, 1 - custom ring tone 1, 2 - custom ring tone 2, 3 - custom ring tone 3. -->
<P582>0</P582>
<!-- Send Anonymous (caller ID will be blocked if set to Yes). 0 - no, 1 - yes -->
<P521>0</P521>
<!-- Anonymous Call Rejection: 0 - no (default), 1 - yes -->
<P546>0</P546>
<!-- Check SIP User ID for incoming INVITE, 0 - no (default), 1 - yes -->
<P549>0</P549>
<!-- Refer-To Uses Target Contact: 0 - no (default), 1 - yes -->
<P569>0</P569>
<!-- Disable Multiple Media Attribute in SDP, 0 - no (default), 1 - yes -->
<P587>0</P587>
<!-- Auto Answer. 0 - no, 1 - yes, 2 - Intercom/Paging -->
<P525>0</P525>
<!-- Preferred Vocoder -->
<!-- 0 - PCMU, 2 - G.726-32, 3 - GSM, 4 - G.723.1, 8 - PCMA, -->
<!-- 9 - G.722, 18 - G.729A/B -->
<!-- Currently only PCMU/PCMA, GSM, G.723, G.729 and G.726-32 supported, will add others in the future -->
<!-- First codec. -->
<P551>0</P551>
<!-- Second codec. -->
<P552>8</P552>
<!-- Third codec. -->
<P553>4</P553>
<!-- Forth codec. -->
<P554>18</P554>
<!-- Fifth codec. -->
<P555>3</P555>
<!-- Sixth codec. -->
<P556>2</P556>
<!-- Seventh codec. -->
<P557>0</P557>
<!-- Eighth codec. -->
<P558>0</P558>
<!-- Preferred Video Codec: 99 - H.264, 34 - H.263, 103 - H.263+ (1998) -->
<!-- Choice 1 -->
<P564>99</P564>
<!-- Choice 2 -->
<P565>34</P565>
<!-- Choice 3 -->
<P575>103</P575>
<!-- Choose Video Codec By Local Preference. 0 - No, 1 - Yes -->
<P574>0</P574>
<!-- Jitter Delay: 0 - Medium, 1 - Low, 2 - High -->
<P567>0</P567>
<!-- Enable Video: 0 - No, 1 - Yes, 2 - No, but allow in-call enabling -->
<P561>1</P561>
<!-- H.264 payload type: between 96 ~ 127, default = 99 -->
<P562>99</P562>
<!-- H.263+ (1998) payload type: between 96 ~ 127, default = 103 -->
<P573>103</P573>
<!-- H.263 Default Resolution. 0 - CIF, 1 - QCIF -->
<P577>0</P577>
<!-- Enable RFC5168 Support. 0 - No, 1 - Yes -->
<P578>0</P578>
<!-- SRTP Mode: Disabled - 0, Enabled but not forced - 1, Enabled and forced - 2 -->
<P543>0</P543>
<!-- Special Feature. 100 - Standard, 101 - Nortel MCS, 102 - BroadSoft, 104 - Sonus, 108 - CBCOM, 109 - RNK -->
<P524>100</P524>
<!-- End User Settings. Please do not edit this section. -->
<!-- Web Access. 0 - HTTP, 1 - HTTPS -->
<!-- P900 = 0 -->
<!-- Web Port: HTTP default is 80 and HTTPS default is 443 -->
<!-- P901 = 80 -->
<!-- End User Password -->
<!-- P196 = 123 -->
<!-- DHCP/PPPoE support. 0 - yes, 1 - no -->
<!-- P8 = 0 -->
<!-- DHCP hostname, alphabet, max. length is 32 -->
<!-- P146 = -->
<!-- DHCP domain, alphabet, max. length is 32 -->
<!-- P147 = -->
<!-- DHCP vendor class ID, alphabet, max. length is 32 -->
<!-- P148 = Grandstream GXV-3000 -->
<!-- PPPoE support. PPPoE user ID -->
<!-- P82 = -->
<!-- PPPoE password -->
<!-- P83 = -->
<!-- PPPoE service name, max. length is 64 alpabit -->
<!-- P269 = -->
<!-- Preferred DNS server, four field, octet digits -->
<!-- P92 = -->
<!-- P93 = -->
<!-- P94 = -->
<!-- P95 = -->
<!-- Static IP Address. Ignore if DHCP or PPPoE is used -->
<!--P9 = 192 -->
<!--P10 = 168 -->
<!--P11 = 0 -->
<!--P12 = 1 -->
<!-- Subnet mask. Ignore if DHCP or PPPoE is used -->
<!--P13 = 255 -->
<!--P14 = 255 -->
<!--P15 = 255 -->
<!--P16 = 0 -->
<!-- Default Router. Ignore if DHCP or PPPoE is used -->
<!--P17 = 192 -->
<!--P18 = 168 -->
<!--P19 = 1 -->
<!--P20 = 1 -->
<!-- DNS 1. Ignore if DHCP or PPPoE is used -->
<!--P21 = 192 -->
<!--P22 = 168 -->
<!--P23 = 0 -->
<!--P24 = 1 -->
<!-- DNS 2. Ignore if DHCP or PPPoE is used -->
<!--P25 = 0 -->
<!--P26 = 0 -->
<!--P27 = 0 -->
<!--P28 = 0 -->
<!-- End User Time settings -->
<!-- Time Zone. Offset in minutes to GMT. -->
<!-- option value = 0 GMT-12:00 (International Date Line West) -->
<!-- option value = 60 GMT-11:00 (Midway Island, Samoa) -->
<!-- option value = 120 GMT-10:00 (US Hawaiian Time) -->
<!-- option value = 180 GMT-9:00 (US Alaska Time) -->
<!-- option value = 240 GMT-8:00 (US Pacific Time, Los Angeles) -->
<!-- option value = 300 GMT-7:00 (US Mountain Time, Denver) -->
<!-- option value = 360 GMT-6:00 (US Central Time, Chicago) -->
<!-- option value = 420 GMT-5:00 (US Eastern Time, New York), Default -->
<!-- option value = 450 GMT-4:30 (Venezuela) -->
<!-- option value = 480 GMT-4:00 (Atlantic Time, Quebec) -->
<!-- option value = 540 GMT-3:00 (Greenland) -->
<!-- option value = 600 GMT-2:00 (Mid-Atlantic) -->
<!-- option value = 660 GMT-1:00 (Azores, Cape Verdi Is.) -->
<!-- option value = 720 GMT 0:00 (London, Dublin, Edinburgh, Lisbon, Casablanca, Monrovia) -->
<!-- option value = 780 GMT+1:00 (Paris, Amsterdam, Berlin, Rome, Vienna, Madrid, Warsaw, Brussels) -->
<!-- option value = 840 GMT+2:00 (Israel, Cairo, Athens, Helsinki, Istanbul, Bucuresti) -->
<!-- option value = 900 GMT+3:00 (Moscow, Kuwait, Baghdad, Nairobi) -->
<!-- option value = 930 GMT+3:30 (Tehran) -->
<!-- option value = 960 GMT+4:00 (Abu Dhabi, Baku) -->
<!-- option value = 990 GMT+4:30 (Kabul) -->
<!-- option value = 1020 GMT+5:00 (Islamabad, Ekaterinburg, Karachi, Tashkent) -->
<!-- option value = 1050 GMT+5:30 (Calcutta, Chennai, Mumbai, New Delhi) -->
<!-- option value = 1065 GMT+5:45 (Kathmandu) -->
<!-- option value = 1080 GMT+6:00 (Almaty, Astana, Dhaka, Novosibirsk) -->
<!-- option value = 1110 GMT+6:30 (Rangoon) -->
<!-- option value = 1140 GMT+7:00 (Bankok, Jakarta, Hanoi, Krasnoyarsk) -->
<!-- option value = 1200 GMT+8:00 (Beijing, Singapore, Taipei, Kuala Lumpur, Irkutsk, Perth) -->
<!-- option value = 1260 GMT+9:00 (Japan, Korea, Yakutsk) -->
<!-- option value = 1290 GMT+9:30 (Adelaide, Darwin) -->
<!-- option value = 1320 GMT+10:00 (Brisbane, Sydney, Melbourne, Canberra, Guam, Hobart) -->
<!-- option value = 1380 GMT+11:00 (Magadan, Solomon Is., New Caledonia) -->
<!-- option value = 1440 GMT+12:00 (Auckland, Wellington, Fiji) -->
<!-- option value = 1500 GMT+13:00 (Nuku'alofa) -->
<P64>420</P64>
<!-- Allow DHCP Option 2 to override Time Zone setting. 0 - No, 1 - Yes. -->
<!-- When set to Yes(1), it will override the configured Time Zone setting if available. -->
<P143>0</P143>
<!-- Daylight Savings Time. 0 - no, 1 - yes -->
<P75>0</P75>
<!-- Optional Rule: -->
<!-- If Daylight Saving Time is selected (P75 = 1), optional rule will allow automatically time ajustment based on the configured rule -->
<!-- Maxlength = 33, default is North America or US Daylight Saving Time Schecule: value="3,2,7,2,0;11,1,7,2,0;60" -->
<P246>3,2,7,2,0;11,1,7,2,0;60</P246>
<!-- Time Display Format. 0 - 12 Hour, 1 - 24 Hour -->
<P122>0</P122>
<!-- Date Display Format. 0 - Year-Month-Day, 1 - Month-Day-Year, 2 - Day-Month-Year -->
<P102>0</P102>
<!-- End User LCD settings -->
<!-- Disable SIP User Display: 0 - no (Default), 1 - yes -->
<P940>0</P940>
<!-- Disable IP Address Display: 0 - no (Default), 1 - yes -->
<P941>0</P941>
<!-- LCD Screen Saver Start Interval: in seconds, 0 means screen saver is off, max. length is 5 -->
<P913>60</P913>
<!-- LCD Screen Saver Refresh Interval: in seconds. from 1-3600, default is 10, controls the time of screensaver image displayed on LCD. -->
<P937>10</P937>
<!-- LCD Auto Power Off Interval: in seconds, 0 means LCD will be always on, max. length is 5 -->
<P909>0</P909>
<!-- LCD Backlight Brightness: max. length is 5, default is 128 -->
<P910>128</P910>
<!-- LCD Contrast: max. length is 5, default is 128 -->
<P911>128</P911>
<!-- LCD Chroma Saturation: max. length is 5, default is 128 -->
<P912>128</P912>
<!-- LCD Text Color: maxlength=3 size=5 -->
<!-- Red (0-255) -->
<P930>255</P930>
<!-- Green (0-255) -->
<P931>255</P931>
<!-- Blue (0-255) -->
<P932>255</P932>
<!-- OSD Text Color: -->
<!-- value = 1 Blue -->
<!-- value = 2 Black -->
<!-- value = 3 Purple -->
<!-- value = 4 Dark Blue -->
<!-- value = 5 Green -->
<!-- value = 6 Red -->
<!-- value = 7 Olive -->
<!-- value = 8 Light Grey -->
<!-- value = 12 Yellow -->
<!-- value = 15 White -->
<P922>15</P922>
<!-- PIP Position: -->
<!-- value=0 Top Left -->
<!-- value=1 Top Right -->
<!-- value=2 Bottom Left -->
<!-- value=3 Bottom Right -->
<P936>0</P936>
<!-- Starting Video OSD/PIP Mode -->
<!-- value=1 OSD On, PIP On -->
<!-- value=2 OSD On, PIP Reversed -->
<!-- value=3 OSD On, PIP Off -->
<!-- value=4 OSD Off, PIP Off -->
<!-- value=5 OSD Off, PIP On -->
<!-- value=6 OSD Off, PIP Reversed -->
<P1332>1</P1332>
<!-- Camera Zoom Mode: -->
<!-- value = 0 1.0:1 (wide and default) -->
<!-- value = 1 1.1:1 -->
<!-- value = 2 1.2:1 -->
<!-- value = 4 1.4:1 -->
<!-- value = 5 1.5:1 -->
<!-- value = 6 1.6:1 -->
<!-- value = 7 1.7:1 -->
<!-- value = 8 1.8:1 -->
<!-- value = 9 1.9:1 -->
<!-- value = 10 2.0:1 <optical tele) -->
<!-- value = 11 2.2:1 -->
<!-- value = 12 2.4:1 -->
<!-- value = 13 2.6:1 -->
<!-- value = 14 2.8:1 -->
<!-- value = 15 3.0:1 -->
<!-- value = 16 3.2:1 -->
<!-- value = 17 3.4:1 -->
<!-- value = 18 3.6:1 -->
<!-- value = 19 3.8:1 -->
<!-- value = 20 4.0:1 (digital tele) -->
<P914>0 </P914>
<!-- Camera Exposure: Default: 0 = auto -->
<!-- value = 0 Auto -->
<!-- value = 5 Very Bright -->
<!-- value = 2 Bright -->
<!-- value = 6 Less Bright I -->
<!-- value = 7 Less Bright II -->
<!-- value = 8 Less Bright II -->
<!-- value = 1 Normal -->
<!-- value = 9 Less Dim I -->
<!-- value = 10 Less Dim II -->
<!-- value = 11 Less Dim III -->
<!-- value = 3 Dim -->
<!-- value = 12 Less Dark -->
<!-- value = 4 Dark -->
<!-- value = 13 Very Dark -->
<P915>0</P915>
<!-- Camera Color Mode: 0 = Color mode (Default), 1 = Monochrome -->
<P919>0</P919>
<!-- Camera White Balance: 0 = auto (Default), 1 = fixed -->
<P920>0</P920>
<!-- Camera Lens Correction: 0 = no, 1 = yes (Default) -->
<P926>1</P926>
<!-- TV Output: 0 = off (Default) -->
<!-- value = 0 Off -->
<!-- value = 1 NTSC -->
<!-- value = 2 PAL-BDGHI -->
<!-- value = 3 PAL-M/PALSA -->
<!-- value = 4 PAL-Nc/PALSA -->
<P926>0</P926>
<!-- FXO Settings. The following settings are only appliable on GXV3005 -->
<!-- Current Disconnect(0, 50 ~ 800ms, default is 100. Enter 0 to disable it.) -->
<P1313>100</P1313>
<!-- Enable Tone Disconnect 0 = No, 1 = Yes -->
<P1314>1</P1314>
<!-- AC Termination Impedance Default: 0 = 600 Ohm (North America) -->
<!-- value = 0 600 Ohm (North America) -->
<!-- value = 1 900 Ohm -->
<!-- value = 2 270 Ohm + (750 Ohm || 150 nF) and 275 Ohm + (780 Ohm || 150 nF) -->
<!-- value = 3 220 Ohm + (820 Ohm || 120 nF) and 220 Ohm + (820 Ohm || 115 nF) -->
<!-- value = 4 370 Ohm + (620 Ohm || 310 nF) -->
<!-- value = 5 320 Ohm + (1050 Ohm || 230 nF) -->
<!-- value = 6 370 Ohm + (820 Ohm || 110 nF) -->
<!-- value = 7 275 Ohm + (780 Ohm || 150 nF) -->
<!-- value = 8 120 Ohm + (820 Ohm || 110 nF) -->
<!-- value = 9 350 Ohm + (1000 Ohm || 210 nF) -->
<!-- value = 10 0 Ohm + (900 Ohm || 30 nF) -->
<!-- value = 11 600 Ohm + 2.16 uF -->
<!-- value = 12 900 Ohm + 1 uF -->
<!-- value = 13 900 Ohm + 2.16 uF -->
<!-- value = 14 600 Ohm + 1 uF -->
<!-- value = 15 Global complex impedance -->
<P1316>0</P1316>
<!-- Silence Timeout (in seconds, default is 60) -->
<P1317>60</P1317>
<!-- Caller ID Scheme (default is BELLCORE) -->
<!-- value = 1 BELLCORE -->
<!-- value = 2 ETSI_PR -->
<!-- value = 3 ETSI_DTAS -->
<!-- value = 4 DTMF -->
<!-- value = 5 NTT -->
<P1318>1</P1318>
<!-- Call Progress Tone -->
<!-- Dial Tone -->
<P1319>f1=350,f2=440;</P1319>
<!-- Ring Back Tone -->
<P1320>f1=440,f2=480,c=200/400;</P1320>
<!-- Busy Tone -->
<P1321>f1=450,f2=450,c=50/50;</P1321>
<!-- Reorder Tone -->
<P1322>f1=480,f2=620,c=25/25;</P1322>
<!-- Tx to PSTN Audio Gain(dB) (-12 ~ 12, default is 1) -->
<P1323>1</P1323>
<!-- Rx to PSTN Audio Gain(dB) (-12 ~ 12, default is 0) -->
<P1324>0</P1324>
<!--FXS Settings. The following settings are only appliable on GXV3006 -->
<!-- Onhook Threshold (Default is 800 ms) -->
<!-- value = 0 Hookflash OFF -->
<!-- value = 2 200 ms -->
<!-- value = 4 400 ms -->
<!-- value = 6 600 ms -->
<!-- value = 8 800 ms -->
<!-- value = 10 1000 ms -->
<!-- value = 12 1200 ms -->
<P245>8</P245>
<!-- Onhook Voltage (Default is 800 ms) -->
<!-- value = 0 48 V -->
<!-- value = 1 18 V -->
<!-- value = 2 24 V -->
<!-- value = 3 36 V -->
<!-- value = 4 51 V -->
<P206>0</P206>
<!-- Volume Amplification -->
<!-- TX (Default is 0 dB) -->
<!-- value = 0 0dB -->
<!-- value = 1 +6dB -->
<!-- value = 2 +4dB -->
<!-- value = 3 +2dB -->
<!-- value = 4 -2dB -->
<!-- value = 5 -4dB -->
<!-- value = 6 -6dB -->
<P247>0</P247>
<!-- RX (Default is 0 dB) -->
<!-- value = 0 0dB -->
<!-- value = 1 +6dB -->
<!-- value = 2 +4dB -->
<!-- value = 3 +2dB -->
<!-- value = 4 -2dB -->
<!-- value = 5 -4dB -->
<!-- value = 6 -6dB -->
<P249>0</P249>
<!-- Caller ID Scheme (default is BELLCORE) -->
<!-- value = 0 BELLCORE -->
<!-- value = 5 SWEDEN_DTMF -->
<P853>0</P853>
<!-- Disable FXS Caller ID 0 = No, 1 = Yes -->
<P1337>0</P1337>
<!-- Enable FXS Call Waiting Caller ID 0 = No, 1 = Yes -->
<P1338>0</P1338>
<!-- Distinctive Ring Tone -->
<!-- system ring tone (f1: 20-50 HZ) -->
<P1333>f1=25,c=200/400;</P1333>
<!-- custom ring tone 1 -->
<P1334>f1=25,c=100/400</P1334>
<!-- custom ring tone 2 -->
<P1335>f1=25,c=100/100-100/300;</P1335>
<!-- custom ring tone 3 -->
<P1336>f1=25,c=100/100-40/50;</P1336>
<!--Account1 -->
<!--audio call, 0=system ring tone(default), 1=custom ring tone 1, 2=custom ring tone 2, 3=custom ring tone 3 -->
<P1355>0</P1355>
<!--video call, , 0=system ring tone(default), 1=custom ring tone 1, 2=custom ring tone 2, 3=custom ring tone 3 -->
<P1356>0</P1356>
<!--Account2 -->
<!--audio call, 0=system ring tone(default), 1=custom ring tone 1, 2=custom ring tone 2, 3=custom ring tone 3 -->
<P483>0</P483>
<!--video call, , 0=system ring tone(default), 1=custom ring tone 1, 2=custom ring tone 2, 3=custom ring tone 3 -->
<P484>0</P484>
<!--Account3 -->
<!--audio call, 0=system ring tone(default), 1=custom ring tone 1, 2=custom ring tone 2, 3=custom ring tone 3 -->
<P484>0</P484>
<!--video call, , 0=system ring tone(default), 1=custom ring tone 1, 2=custom ring tone 2, 3=custom ring tone 3 -->
<P584>0</P584>
</config>
</gs_provision>

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<?xml version="1.0" encoding="UTF-8" ?>
<gs_provision version="1">
<config version="1">
<!-- Advanced Settings -->
<!-- Admin password for web interface,String Max Length: 30; between ascii values 33 and 126, Mandatory -->
<P2>admin</P2>
<!-- Firmware Upgrade -->
<!-- Firmware Upgrade and Privisioning. 0 - TFTP Upgrade, 1 - HTTP Upgrade, 2 - HTTPS Upgrade., Number: 0 to 2. Mandatory -->
<P212>1</P212>
<!-- Config Server Path , String: serveraddress -->
<P237>{$domain_name}{$project_path}/app/provision</P237>
<!-- Profile 1 Settings -->
<!-- Profile Active. 0 - no, 1 - yes, Number: 0,1. -->
<P271>1</P271>
<!-- Primary SIP Server,String: serveraddress -->
<P47>{$server_address_1}</P47>
<!-- Prefer Primary SIP Server. 0 - No, 1 - Yes. -->
<P4567>1</P4567>
<!-- SIP Transport. 0 - UDP, 1 - TCP, 2 - TLS -->
<P130>0</P130>
<!-- NAT Traversal (STUN). 0 - No, 2 - No but send keep-alive, 1 - Yes -->
<P52>0</P52>
<!-- SIP Registration. 0 - no, 1 - yes -->
<P31>1</P31>
<!-- Register Expiration (in minutes. default 1 hour, max 45 days), Number: 1 to 64800 -->
<P32>3</P32>
<!--FXS Ports -->
<!-- FXS Port 1,SIP USER ID, Authenticate ID, Password, Name, Profile ID (0 - Profile 1, 1 - Profile 2, 2 - Profile 3) -->
<!-- Hunting Group (0 - None, 1 - Active, 2 - 2, 3 - 3, 4 -4, 5 - 5, 6 - 6, 7 - 7, 8 - 8 ) -->
<!-- Request URI Routing ID, Enable Port(0 - No, 1 - Yes, default is Yes) -->
<P4060>{$user_id_1}</P4060>
<P4090>{$user_id_1}</P4090>
<P4120>{$user_password_1}</P4120>
<P4180>{$display_name_1}</P4180>
<P4150>0</P4150>
<P4300>0</P4300>
<P4669></P4669>
<P4595>1</P4595>
<!-- FXS Port 2. SIP USER ID, Authenticate ID, Password, Name, Profile ID (0 - Profile 1, 1 - Profile 2, 2 - Profile 3) -->
<!-- Hunting Group (0 - None, 2 - Active, 1 - 1, 3 - 3, 4 -4, 5 - 5, 6 - 6, 7 - 7, 8 - 8 ) -->
<!-- Request URI Routing ID.Enable Port(0 - No, 1 - Yes, default is Yes) -->
<P4061>{$user_id_2}</P4061>
<P4091>{$user_id_2}</P4091>
<P4121>{$user_password_2}</P4121>
<P4181>{$display_name_2}</P4181>
<P4151>0</P4151>
<P4301>0</P4301>
<P4670></P4670>
<P4596>1</P4596>
<!-- FXS Port 3. SIP USER ID, Authenticate ID, Password, Name, Profile ID (0 - Profile 1, 1 - Profile 2, 2 - Profile 3) -->
<!-- Hunting Group (0 - None, 3 - Active, 1 - 1, 2 - 2, 4 -4, 5 - 5, 6 - 6, 7 - 7, 8 - 8 ) -->
<!-- Request URI Routing ID. Enable Port(0 - No, 1 - Yes, default is Yes) -->
<P4062>{$user_id_3}</P4062>
<P4092>{$user_id_3}</P4092>
<P4122>{$user_password_3}</P4122>
<P4182>{$display_name_3}</P4182>
<P4152>0</P4152>
<P4302>0</P4302>
<P4671></P4671>
<P4597>1</P4597>
<!-- FXS Port 4. SIP USER ID, Authenticate ID, Password, Name, Profile ID (0 - Profile 1, 1 - Profile 2, 2 - Profile 3) -->
<!-- Hunting Group (0 - None, 4 - Active, 1 - 1, 2 - 2, 3 - 3, 5 - 5, 6 - 6, 7 - 7, 8 - 8 ) -->
<!-- Request URI Routing ID. Enable Port(0 - No, 1 - Yes, default is Yes) -->
<P4063>{$user_id_4}</P4063>
<P4093>{$user_id_4}</P4093>
<P4123>{$user_password_4}</P4123>
<P4183>{$display_name_4}</P4183>
<P4153>0</P4153>
<P4303>0</P4303>
<P4672></P4672>
<P4598>1</P4598>
</config>
</gs_provision>

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<?xml version="1.0" encoding="UTF-8" ?>
<gs_provision version="1">
<config version="1">
<!-- Advanced Settings -->
<!-- Admin password for web interface,String Max Length: 30; between ascii values 33 and 126, Mandatory -->
<P2>admin</P2>
<!-- Firmware Upgrade -->
<!-- Firmware Upgrade and Privisioning. 0 - TFTP Upgrade, 1 - HTTP Upgrade, 2 - HTTPS Upgrade., Number: 0 to 2. Mandatory -->
<P212>1</P212>
<!-- Config Server Path , String: serveraddress -->
<P237>{$domain_name}{$project_path}/app/provision</P237>
<!-- Profile 1 Settings -->
<!-- Profile Active. 0 - no, 1 - yes, Number: 0,1. -->
<P271>1</P271>
<!-- Primary SIP Server,String: serveraddress -->
<P47>{$server_address_1}</P47>
<!-- Prefer Primary SIP Server. 0 - No, 1 - Yes. -->
<P4567>1</P4567>
<!-- SIP Transport. 0 - UDP, 1 - TCP, 2 - TLS -->
<P130>0</P130>
<!-- NAT Traversal (STUN). 0 - No, 2 - No but send keep-alive, 1 - Yes -->
<P52>0</P52>
<!-- SIP Registration. 0 - no, 1 - yes -->
<P31>1</P31>
<!-- Register Expiration (in minutes. default 1 hour, max 45 days), Number: 1 to 64800 -->
<P32>3</P32>
<!--FXS Ports -->
<!-- FXS Port 1,SIP USER ID, Authenticate ID, Password, Name, Profile ID (0 - Profile 1, 1 - Profile 2, 2 - Profile 3) -->
<!-- Hunting Group (0 - None, 1 - Active, 2 - 2, 3 - 3, 4 -4, 5 - 5, 6 - 6, 7 - 7, 8 - 8 ) -->
<!-- Request URI Routing ID, Enable Port(0 - No, 1 - Yes, default is Yes) -->
<P4060>{$user_id_1}</P4060>
<P4090>{$user_id_1}</P4090>
<P4120>{$user_password_1}</P4120>
<P4180>{$display_name_1}</P4180>
<P4150>0</P4150>
<P4300>0</P4300>
<P4669></P4669>
<P4595>1</P4595>
<!-- FXS Port 2. SIP USER ID, Authenticate ID, Password, Name, Profile ID (0 - Profile 1, 1 - Profile 2, 2 - Profile 3) -->
<!-- Hunting Group (0 - None, 2 - Active, 1 - 1, 3 - 3, 4 -4, 5 - 5, 6 - 6, 7 - 7, 8 - 8 ) -->
<!-- Request URI Routing ID.Enable Port(0 - No, 1 - Yes, default is Yes) -->
<P4061>{$user_id_2}</P4061>
<P4091>{$user_id_2}</P4091>
<P4121>{$user_password_2}</P4121>
<P4181>{$display_name_2}</P4181>
<P4151>0</P4151>
<P4301>0</P4301>
<P4670></P4670>
<P4596>1</P4596>
<!-- FXS Port 3. SIP USER ID, Authenticate ID, Password, Name, Profile ID (0 - Profile 1, 1 - Profile 2, 2 - Profile 3) -->
<!-- Hunting Group (0 - None, 3 - Active, 1 - 1, 2 - 2, 4 -4, 5 - 5, 6 - 6, 7 - 7, 8 - 8 ) -->
<!-- Request URI Routing ID. Enable Port(0 - No, 1 - Yes, default is Yes) -->
<P4062>{$user_id_3}</P4062>
<P4092>{$user_id_3}</P4092>
<P4122>{$user_password_3}</P4122>
<P4182>{$display_name_3}</P4182>
<P4152>0</P4152>
<P4302>0</P4302>
<P4671></P4671>
<P4597>1</P4597>
<!-- FXS Port 4. SIP USER ID, Authenticate ID, Password, Name, Profile ID (0 - Profile 1, 1 - Profile 2, 2 - Profile 3) -->
<!-- Hunting Group (0 - None, 4 - Active, 1 - 1, 2 - 2, 3 - 3, 5 - 5, 6 - 6, 7 - 7, 8 - 8 ) -->
<!-- Request URI Routing ID. Enable Port(0 - No, 1 - Yes, default is Yes) -->
<P4063>{$user_id_4}</P4063>
<P4093>{$user_id_4}</P4093>
<P4123>{$user_password_4}</P4123>
<P4183>{$display_name_4}</P4183>
<P4153>0</P4153>
<P4303>0</P4303>
<P4672></P4672>
<P4598>1</P4598>
<!-- The following ports (5 to 8) belong to GXW-4008/GXW-4024 only -->
<!-- FXS Port 5. SIP USER ID, Authenticate ID, Password, Name, Profile ID (0 - Profile 1, 1 - Profile 2, 2 - Profile 3) -->
<!-- Hunting Group (0 - None, 5 - Active, 1 - 1, 2 - 2, 3 - 3, 4 -4, 6 - 6, 7 - 7, 8 - 8 ) -->
<!-- Request URI Routing ID. Enable Port(0 - No, 1 - Yes, default is Yes) -->
<P4064>{$user_id_5}</P4064>
<P4094>{$user_id_5}</P4094>
<P4124>{$user_password_5}</P4124>
<P4184>{$display_name_5}</P4184>
<P4154>0</P4154>
<P4304>0</P4304>
<P4673></P4673>
<P4599>1</P4599>
<!-- FXS Port 6. SIP USER ID, Authenticate ID, Password, Name, Profile ID (0 - Profile 1, 1 - Profile 2, 2 - Profile 3) -->
<!-- Hunting Group (0 - None, 6 - Active, 1 - 1, 2 - 2, 3 - 3, 4 -4, 5 - 5, 7 - 7, 8 - 8 ) -->
<!-- Request URI Routing ID. Enable Port(0 - No, 1 - Yes, default is Yes) -->
<P4065>{$user_id_6}</P4065>
<P4095>{$user_id_6}</P4095>
<P4125>{$user_password_6}</P4125>
<P4185>{$display_name_6}</P4185>
<P4155>0</P4155>
<P4305>0</P4305>
<P4674></P4674>
<P4600>1</P4600>
<!-- FXS Port 7. SIP USER ID, Authenticate ID, Password, Name, Profile ID (0 - Profile 1, 1 - Profile 2, 2 - Profile 3) -->
<!-- Hunting Group (0 - None, 7 - Active, 1 - 1, 2 - 2, 3 - 3, 4 -4, 5 - 5, 6 - 6, 8 - 8 ) -->
<!-- Request URI Routing ID. Enable Port(0 - No, 1 - Yes, default is Yes) -->
<P4066>{$user_id_7}</P4066>
<P4096>{$user_id_7}</P4096>
<P4126>{$user_password_7}</P4126>
<P4186>{$display_name_7}</P4186>
<P4156>0</P4156>
<P4306>0</P4306>
<P4675></P4675>
<P4601>1</P4601>
<!-- FXS Port 8. SIP USER ID, Authenticate ID, Password, Name, Profile ID (0 - Profile 1, 1 - Profile 2, 2 - Profile 3) -->
<!-- Hunting Group (0 - None, 8 - Active, 1 - 1, 2 - 2, 3 - 3, 4 -4, 5 - 5, 6 - 6, 7 - 7 ) -->
<!-- Request URI Routing ID. Enable Port(0 - No, 1 - Yes, default is Yes) -->
<P4067>{$user_id_8}</P4067>
<P4097>{$user_id_8}</P4097>
<P4127>{$user_password_8}</P4127>
<P4187>{$display_name_8}</P4187>
<P4157>0</P4157>
<P4307>0</P4307>
<P4676></P4676>
<P4602>1</P4602>
</config>
</gs_provision>

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<?xml version="1.0" encoding="utf-8"?>
<gs_provision version="1">
<config version="1">
<!-- Configuration template for GXW410x version 1.3.4.13 -->
<!-- Advanced/System-wide Options -->
<!-- Admin password for web interface, max. length is 30 -->
<P2>admin</P2>
<!-- G.723 rate: 0 - 6.3kbps encoding rate, 1 - 5.3kbps encoding rate -->
<P49>0</P49>
<!-- Voice Frames per TX (up to 10/20/32/64 frames for G711/G726/G723/other codecs respectively) -->
<P37>2</P37>
<!-- Layer 3 QoS (IP Diff-Serv or Precedence value for RTP) -->
<P38>48</P38>
<!-- Layer 2 QoS. 802.1Q/VLAN Tag (VLAN classification for RTP) -->
<P51>0</P51>
<!-- Layer 2 QoS. 802.1p priority value (0 - 7) -->
<P87>0</P87>
<!-- Local RTP port (1024-65535, default 5004) -->
<P39>5004</P39>
<!-- Use Random Port. 0 - no, 1 - yes -->
<P78>0</P78>
<!-- Keep-alive interval (in seconds. default 20 seconds) -->
<P84>20</P84>
<!-- Use NAT IP. This will enable our SIP client to use this IP in the SIP message. Example 64.3.153.50. -->
<P101></P101>
<!-- STUN server -->
<P76></P76>
<!-- Firmware Upgrade -->
<!-- Firmware Upgrade. 0 - TFTP Upgrade, 1 - HTTP Upgrade. -->
<P212>1</P212>
<!-- Firmware Server Path -->
<P192>fm.grandstream.com/gs</P192>
<!-- Config Server Path -->
<P237>{$domain_name}{$project_path}/app/provision</P237>
<!-- Firmware File Prefix -->
<P232></P232>
<!-- Firmware File Postfix -->
<P233></P233>
<!-- Config File Prefix -->
<P234></P234>
<!-- Config File Postfix -->
<P235></P235>
<!-- Allow DHCP Option 66 to override server. 0 - No, 1 - Yes. Default is Yes. -->
<!-- When set to Yes(1), it will override the configured provision path and method. -->
<P145>1</P145>
<!-- Automatic Upgrade. 0 - No, 1 - Yes. Default is No. -->
<P194>0</P194>
<!-- Check for new firmware every () minutes, unit is in minute, minimnu 60 minutes, default is 7 days. -->
<P193>10080</P193>
<!-- Use firmware pre/postfix to determine if f/w is required -->
<!-- 0 = Always Check for New Firmware -->
<!-- 1 = Check New Firmware only when F/W pre/suffix changes -->
<!-- 2 = Always Skip the Firmware Check -->
<P238>0</P238>
<!-- DTMF Payload Type -->
<P79>101</P79>
<!-- Syslog Server (name of the server, max length is 64 charactors) -->
<P207></P207>
<!-- Syslog Level (Default setting is NONE) -->
<!-- 0 - NONE, 1 - DEBUG, 2 - INFO, 3 - WARNING, 4 - ERROR -->
<P208>0</P208>
<!-- NTP Server -->
<P30>us.pool.ntp.org</P30>
<!-- Allow DHCP Option 42 to override NTP server. 0 - No, 1 - Yes. Default is No. -->
<!-- When set to Yes(1), it will override the configured NTP server. -->
<P144>0</P144>
<!-- Enable Video Surveillance -->
<!-- 0 - No, 1 - Yes -->
<P928>0</P928>
<!-- RTSP Port -->
<P929>554</P929>
<!-- RTP Loopback. 0 - No, 1 - Yes. (No as default, Yes means no RTP if RTP streams between 2 internal ports) -->
<!-- P3598 = 0 -->
<!-- Profile 1 Settings -->
<!-- Activate Profile 0 - no, 1 - yes -->
<P271>1</P271>
<!-- Profile Name -->
<P270></P270>
<!-- SIP Server: max. length = 96 -->
<P47>{$server_address_1}</P47>
<!-- Outbound Proxy -->
<P48></P48>
<!-- Use DNS SRV. 0 - No, 1 - Yes. -->
<P103>0</P103>
<!-- SIP User ID is phone number. 0 - no, 1 - yes -->
<P63>0</P63>
<!-- SIP Registration. 0 - no, 1 - yes -->
<P31>1</P31>
<!-- Unregister On Reboot. 0 - no, 1 - yes -->
<P81>0</P81>
<!-- Register Expiration (in minutes. default 1 hour, max 45 days) -->
<P32>3</P32>
<!-- Registration Failure Retry Wait Time (in seconds. Between 1-3600, default is 20) -->
<P138>20</P138>
<!-- SIP Transport. 1 - UDP, 2 - TCP , (default is UDP) -->
<P130>1</P130>
<!-- NAT Traversal. 0 - yes, 1 - no, 2 - No, but send keep-alive -->
<P52>1</P52>
<!-- Proxy-Require (A SIP extension to enable firewall penetration) -->
<P197></P197>
<!-- Early Dial (use "Yes" only if proxy supports 484 response). 0 - no, 1 - yes -->
<P29>0</P29>
<!-- Session Expiration (in seconds. default 180 seconds. Allowed value: 90-65535) -->
<P260>180</P260>
<!-- Minimum SE (in seconds. default 90 seconds, must be lower than or equal to P260) -->
<P261>90</P261>
<!-- Caller Request Timer (Request for timer when calling) 0 - no, 1 - yes -->
<P262>0</P262>
<!-- Callee Request Timer (Request for timer when called. i.e. if remote party supports timer but did not request for one) 0 - no, 1 - yes -->
<P263>0</P263>
<!-- Force Timer (Still use timer when remote party does not support timer) 0 - no, 1 - yes -->
<P264>0</P264>
<!-- Force INVITE (Always refresh with INVITE instead of UPDATE even when remote party supports UPDATE) 0 - no, 1 - yes -->
<P265>0</P265>
<!-- UAC Specify Refresher. 0 - omit, 1 - UAC, 2 - UAS -->
<P266>0</P266>
<!-- UAS Specify Refresher. 1 - UAC, 2 - UAS -->
<P267>1</P267>
<!-- Enable 100rel . 0 - no, 1 - yes -->
<P272>0</P272>
<!-- Refer to Uses Target Contact -->
<!-- 0 - No, 1 - Yes -->
<P135>0</P135>
<!-- INVITE Ring-no-anwser Timeout -->
<P3180>40</P3180>
<!-- Accept INVITE from Proxy Only (100 %u2013No; 101-YES, default YES) -->
<P743>101</P743>
<!-- Codec/Voice Quality settings -->
<!-- First codec. 0 - PCMU, 3 - GSM, 4 - G.723.1, 8 - PCMA, 18 - G.729A/B -->
<P57>0</P57>
<!-- Second codec. 0 - PCMU, 3 - GSM, 4 - G.723.1, 8 - PCMA, 18 - G.729A/B -->
<P58>3</P58>
<!-- Third codec. 0 - PCMU, 3 - GSM, 4 - G.723.1, 8 - PCMA, 18 - G.729A/B -->
<P59>4</P59>
<!-- Forth codec. 0 - PCMU, 3 - GSM, 4 - G.723.1, 8 - PCMA, 18 - G.729A/B -->
<P60>8</P60>
<!-- Fifth codec. 0 - PCMU, 3 - GSM, 4 - G.723.1, 8 - PCMA, 18 - G.729A/B -->
<P61>18</P61>
<!-- Sixth codec. 0 - PCMU, 3 - GSM, 4 - G.723.1, 8 - PCMA, 18 - G.729A/B -->
<P62>0</P62>
<!-- Seventh codec. 0 - PCMU, 3 - GSM, 4 - G.723.1, 8 - PCMA, 18 - G.729A/B -->
<P46>0</P46>
<!-- Eighth codec. 0 - PCMU, 3 - GSM, 4 - G.723.1, 8 - PCMA, 18 - G.729A/B -->
<P98>0</P98>
<!-- Special Feature. 100 - Standard, 101 - Nortel MCS, 102 - BroadSoft, 104 - Sonus, 108 - CBCOM, 109 - RNK -->
<P198>100</P198>
<!-- Profile 2 Settings -->
<!-- Activate Profile 0 - no, 1 - yes -->
<P401>0</P401>
<!-- Profile Name -->
<P417></P417>
<!-- SIP Server -->
<P402>{$server_address_2}</P402>
<!-- Outbound Proxy Server -->
<P403></P403>
<!-- Use DNS SRV. 0 - No, 1 - Yes. -->
<P408>0</P408>
<!-- User ID is phone number; "user=phone". 0 - no, 1 - yes -->
<P409>0</P409>
<!-- SIP Registration. 0 - no, 1 - yes -->
<P410>1</P410>
<!-- Unregister on Reboot. 0 - no, 1 - yes -->
<P411>0</P411>
<!-- Register Expiration (in minutes. default 1 hour, max 45 days) -->
<P412>3</P412>
<!-- Registration Failure Retry Wait Time (in seconds. Between 1-3600, default is 20) -->
<P471>20</P471>
<!-- SIP Transport. 1 - UDP, 2 - TCP , (default is UDP) -->
<P448>1</P448>
<!-- NAT Traversal. 0 - yes, 1 - no, 2 - No, but send keep-alive -->
<P414>1</P414>
<!-- Proxy Require -->
<P418></P418>
<!-- Early Dial. 0 - no, 1 - yes -->
<P422>0</P422>
<!-- Session Expiration (in seconds. default 180 seconds. Allowed value: 90-65535) -->
<P434>180</P434>
<!-- Minimum SE (in seconds. default 90 seconds, must be lower than or equal to P260) -->
<P427>90</P427>
<!-- Caller Request Timer (Request for timer when calling) 0 - no, 1 - yes -->
<P428>0</P428>
<!-- Callee Request Timer (Request for timer when called. i.e. if remote party supports timer but did not request for one) 0 - no, 1 - yes -->
<P429>0</P429>
<!-- Force Timer (Still use timer when remote party does not support timer) 0 - no, 1 - yes -->
<P430>0</P430>
<!-- Force INVITE (Always refresh with INVITE instead of UPDATE even when remote party supports UPDATE) 0 - no, 1 - yes -->
<P431>0</P431>
<!-- UAC Specify Refresher. 0 - omit, 1 - UAC, 2 - UAS -->
<P432>0</P432>
<!-- UAS Specify Refresher. 1 - UAC, 2 - UAS -->
<P433>1</P433>
<!-- Enable 100rel. 0 - no, 1 - yes -->
<P435>0</P435>
<!-- Refer to Uses Target Contact -->
<!-- 0 - No, 1 - Yes -->
<P469>0</P469>
<!-- INVITE Ring-no-answer Timeout -->
<P3181>40</P3181>
<!-- Accept INVITE from Proxy Only (100 %u2013No; 101-YES, default YES) -->
<P4043>101</P4043>
<!-- First codec. 0 - PCMU, 3 - GSM, 4 - G.723.1, 8 - PCMA, 18 - G.729A/B -->
<P451>0</P451>
<!-- Second codec. 0 - PCMU, 3 - GSM, 4 - G.723.1, 8 - PCMA, 18 - G.729A/B -->
<P452>3</P452>
<!-- Third codec. 0 - PCMU, 3 - GSM, 4 - G.723.1, 8 - PCMA, 18 - G.729A/B -->
<P453>4</P453>
<!-- Forth codec. 0 - PCMU, 3 - GSM, 4 - G.723.1, 8 - PCMA, 18 - G.729A/B -->
<P454>8</P454>
<!-- Fifth codec. 0 - PCMU, 3 - GSM, 4 - G.723.1, 8 - PCMA, 18 - G.729A/B -->
<P455>18</P455>
<!-- Sixth codec. 0 - PCMU, 3 - GSM, 4 - G.723.1, 8 - PCMA, 18 - G.729A/B -->
<P456>0</P456>
<!-- Seventh codec. 0 - PCMU, 3 - GSM, 4 - G.723.1, 8 - PCMA, 18 - G.729A/B -->
<P457>0</P457>
<!-- Eighth codec. 0 - PCMU, 3 - GSM, 4 - G.723.1, 8 - PCMA, 18 - G.729A/B -->
<P458>0</P458>
<!-- Special Feature. 100 - Standard, 101 - Nortel MCS, 102 - BroadSoft, 104 - Sonus, 108 - CBCOM, 109 - RNK -->
<P424>100</P424>
<!-- Profile 3 Settings -->
<!-- Activate Profile 0 - no, 1 - yes -->
<P501>0</P501>
<!-- Profile Name -->
<P517></P517>
<!-- SIP Server -->
<P502>{$server_address_3}</P502>
<!-- Outbound Proxy Server -->
<P503></P503>
<!-- Use DNS SRV. 0 - No, 1 - Yes. -->
<P508>0</P508>
<!-- User ID is phone. "user=phone". 0 - no, 1 - yes -->
<P509>0</P509>
<!-- SIP Registration. 0 - no, 1 - yes -->
<P510>1</P510>
<!-- Unregister on Reboot. 0 - no, 1 - yes -->
<P511>0</P511>
<!-- Register Expiration (in minutes. default 1 hour, max 45 days) -->
<P512>3</P512>
<!-- Registration Failure Retry Wait Time (in seconds. Between 1-3600, default is 20) -->
<P571>20</P571>
<!-- SIP Transport. 1 - UDP, 2 - TCP , (default is UDP) -->
<P548>1</P548>
<!-- NAT Traversal. 0 - yes, 1 - no -->
<P514>1</P514>
<!-- Proxy Require -->
<P518></P518>
<!-- Early Dial. 0 - no, 1 - yes -->
<P522>0</P522>
<!-- Session Expiration (in seconds. default 180 seconds. Allowed value: 90-65535) -->
<P534>180</P534>
<!-- Minimum SE (in seconds. default 90 seconds, must be lower than or equal to P260) -->
<P527>90</P527>
<!-- Caller Request Timer (Request for timer when calling) 0 - no, 1 - yes -->
<P528>0</P528>
<!-- Callee Request Timer (Request for timer when called. i.e. if remote party supports timer but did not request for one) 0 - no, 1 - yes -->
<P529>0</P529>
<!-- Force Timer (Still use timer when remote party does not support timer) 0 - no, 1 - yes -->
<P530>0</P530>
<!-- Force INVITE (Always refresh with INVITE instead of UPDATE even when remote party supports UPDATE) 0 - no, 1 - yes -->
<P531>0</P531>
<!-- UAC Specify Refresher. 0 - omit, 1 - UAC, 2 - UAS -->
<P532>0</P532>
<!-- UAS Specify Refresher. 1 - UAC, 2 - UAS -->
<P533>1</P533>
<!-- Enable 100rel. 0 - no, 1 - yes -->
<P535>0</P535>
<!-- Refer to Uses Target Contact -->
<!-- 0 - No, 1 - Yes -->
<P569>0</P569>
<!-- INVITE Ring-no-anwser Timeout -->
<P3182>40</P3182>
<!-- Accept INVITE from Proxy Only (100 %u2013No; 101-YES, default YES) -->
<P4044>101</P4044>
<!-- Codec/Voice Quality settings -->
<!-- First codec. 0 - PCMU, 3 - GSM, 4 - G.723.1, 8 - PCMA, 18 - G.729A/B -->
<P551>0</P551>
<!-- Second codec. 0 - PCMU, 3 - GSM, 4 - G.723.1, 8 - PCMA, 18 - G.729A/B -->
<P552>3</P552>
<!-- Third codec. 0 - PCMU, 3 - GSM, 4 - G.723.1, 8 - PCMA, 18 - G.729A/B -->
<P553>4</P553>
<!-- Forth codec. 0 - PCMU, 3 - GSM, 4 - G.723.1, 8 - PCMA, 18 - G.729A/B -->
<P554>8</P554>
<!-- Fifth codec. 0 - PCMU, 3 - GSM, 4 - G.723.1, 8 - PCMA, 18 - G.729A/B -->
<P555>18</P555>
<!-- Sixth codec. 0 - PCMU, 3 - GSM, 4 - G.723.1, 8 - PCMA, 18 - G.729A/B -->
<P556>0</P556>
<!-- Seventh codec. 0 - PCMU, 3 - GSM, 4 - G.723.1, 8 - PCMA, 18 - G.729A/B -->
<P557>0</P557>
<!-- Eighth codec. 0 - PCMU, 3 - GSM, 4 - G.723.1, 8 - PCMA, 18 - G.729A/B -->
<P558>0</P558>
<!-- Special Feature. 100 - Standard, 101 - Nortel MCS, 102 - BroadSoft, 104 - Sonus, 108 - CBCOM, 109 - RNK -->
<P524>100</P524>
<!-- FXO Lines Settings -->
<!-- FXO Termination -->
<!-- Enable Current Disconnect -->
<!-- N - No, Y - Yes -->
<!-- For GXW4104 (uncomment the following line by removing #) -->
<!-- P3603 = ch1-4:Y; -->
<!-- For GXW4108 (uncomment the following line by removing #) -->
<!-- P3603 = ch1-8:Y; -->
<!-- Enable Current Disconnect Threshold. Default 100, normally 100 ~ 800ms. -->
<!-- For GXW4104 (uncomment the following line by removing #) -->
<!-- P3604 = ch1-4:100; -->
<!-- For GXW4108 (uncomment the following line by removing #) -->
<!-- P3604 = ch1-8:100; -->
<!-- Enable Tone Disconnect. N - No, Y - Yes. -->
<!-- If set yes, reorder tone is used as disconnect signal. Default No. -->
<!-- For GXW4104 (uncomment the following line by removing #) -->
<!-- P3605 = ch1-4:Y; -->
<!-- For GXW4108 (uncomment the following line by removing #) -->
<!-- P3605 = ch1-8:Y; -->
<!-- Enable Polarity Reversal. N - No, Y - Yes. -->
<!-- Default No. Check with your PSTN carrier before set to Yes -->
<!-- For GXW4104 (uncomment the following line by removing #) -->
<!-- P3606 = ch1-4:N; -->
<!-- For GXW4108 (uncomment the following line by removing #) -->
<!-- P3606 = ch1-8:N; -->
<!-- Enable Call Answer Supervision. N - No, Y - Yes. -->
<!-- Default No. Check with your PSTN carrier before set to Yes -->
<!-- For GXW4104 (uncomment the following line by removing #) -->
<!-- P3629 = ch1-4:N; -->
<!-- For GXW4108 (uncomment the following line by removing #) -->
<!-- P3629 = ch1-8:N; -->
<!-- Silence Timeout. Default 60 seconds. -->
<!-- For GXW4104 (uncomment the following line by removing #) -->
<!-- P3202 = ch1-4:60; -->
<!-- For GXW4108 (uncomment the following line by removing #) -->
<!-- P3202 = ch1-8:60; -->
<!-- Incoming Call Ring Timeout. Allowed value: 2-10s, default 6s. -->
<!-- For GXW4104 (uncomment the following line by removing #) -->
<!-- P3630 = ch1-4:6; -->
<!-- For GXW4108 (uncomment the following line by removing #) -->
<!-- P3630 = ch1-8:6; -->
<!-- AC Termination Impedance -->
<!-- 0 - 600 Ohm ( North American ) -->
<!-- 1 - 900 Ohm -->
<!-- 2 - 270 Ohm + (750 Ohm || 150nF) and 275 Ohm + (780 Ohm || 150nF) -->
<!-- 3 - 220 Ohm + (820 Ohm || 120nF) and 220 Ohm + (820 Ohm || 115nF) -->
<!-- 4 - 370 Ohm + (620 Ohm || 310nF) -->
<!-- 5 - 320 Ohm + (1050 Ohm || 230nF) -->
<!-- 6 - 370 Ohm + (820 Ohm || 110nF) -->
<!-- 7 - 275 Ohm + (78 Ohm || 150 nF) -->
<!-- 8 - 120 Ohm + (820 Ohm || 110 nF) -->
<!-- 9 - 350 Ohm + (1000 Ohm || 210nF) -->
<!-- 10 - 0 Ohm + (900 Ohm || 30nF) -->
<!-- 11 - 600 Ohm + 2.16 uF -->
<!-- 12 - 900 Ohm + 1 uF -->
<!-- 13 - 900 Ohm + 2.16 uF -->
<!-- 14 - 600 Ohm + 1 uF -->
<!-- 15 - Global complex impedance -->
<!-- For GXW4104 (uncomment the following line by removing #) -->
<!-- P3631 = ch1-4:0; -->
<!-- For GXW4108 (uncomment the following line by removing #) -->
<!-- P3631 = ch1-8:0; -->
<!-- Channel Dialing to PSTN -->
<!-- Wait for Dial Tone (Y/N). 0 - No, 1 - Yes. Default Yes - dial upon dial-tone -->
<!-- For GXW4104 (uncomment the following line by removing #) -->
<!-- P3303 = ch1-4:N; -->
<!-- For GXW4108 (uncomment the following line by removing #) -->
<!-- P3303 = ch1-8:N; -->
<!-- Stage Method(1/2). Default 2 - 2 stage dialing. -->
<!-- For GXW4104 (uncomment the following line by removing #) -->
<!-- P3304 = ch1-4:2; -->
<!-- For GXW4108 (uncomment the following line by removing #) -->
<!-- P3304 = ch1-8:2; -->
<!-- Min Delay Before Dial PSTN. Default 500, range 50 ~ 6500ms. -->
<!-- For GXW4104 (uncomment the following line by removing #) -->
<!-- P3206 = ch1-4:500; -->
<!-- For GXW4108 (uncomment the following line by removing #) -->
<!-- P3206 = ch1-8:500; -->
<!-- Unconditional Call Forward to VOIP. -->
<!-- User ID -->
<!-- For GXW4104 (uncomment the following line by removing #) -->
<!-- P3607 = ch1-4:; -->
<!-- For GXW4108 (uncomment the following line by removing #) -->
<!-- P3607 = ch1-8:; -->
<!-- SIP Server. -->
<!-- For GXW4104 (uncomment the following line by removing #) -->
<!-- P3306 = ch1-4:p1; -->
<!-- For GXW4108 (uncomment the following line by removing #) -->
<!-- P3306 = ch1-8:p1; -->
<!-- Sip Destination Port. -->
<!-- For GXW4104 (uncomment the following line by removing #) -->
<!-- P3307 = ch1-4:5060; -->
<!-- For GXW4108 (uncomment the following line by removing #) -->
<!-- P3307 = ch1-8:5060; -->
<!-- PSTN to VoIP Caller ID Setting -->
<!-- Number of Rings Before Pickup. Default 4. Allowed values: 1-50. -->
<!-- For GXW4104 (uncomment the following line by removing #) -->
<!-- P3599 = ch1-4:4; -->
<!-- For GXW4108 (uncomment the following line by removing #) -->
<!-- P3599 = ch1-8:4; -->
<!-- Caller ID Scheme -->
<!-- 1:Bellcore/Telcordia, 2:ETSI-FSK during ringing, 3:ETSI-FSK prior to ringing with DTAS -->
<!-- 4:ETSI-FSK prior to ringing with LR, 5:ETSI-FSK prior to ringing with RP, 6:ETSI-DTMF during ringing -->
<!-- 7:ETSI-DTMF prior to ringing with DTAS, 8:ETSI-DTMF prior to ringing with LR -->
<!-- 9:ETSI-DTMF prior to ringing with RP, 10:SIN 227 BT, 11: NTT -->
<!-- For GXW4104 (uncomment the following line by removing #) -->
<!-- P3600 = ch1-4:1; -->
<!-- For GXW4108 (uncomment the following line by removing #) -->
<!-- P3600 = ch1-8:1; -->
<!-- Caller ID Transport Type. Default 1. -->
<!-- 1:Relay via SIP From, 2:Disable, 3:Send Anonymous, 4:Relay via SIP P-Asserted-Identity -->
<!-- For GXW4104 (uncomment the following line by removing #) -->
<!-- P3601 = ch1-4:1; -->
<!-- For GXW4108 (uncomment the following line by removing #) -->
<!-- P3601 = ch1-8:1; -->
<!-- T.38 Setting -->
<!-- Syntax: ch x-y: mode=val,rate=val,ecm=val;[...] -->
<!-- (mode: 1:Relay(default), 2:Passthough) -->
<!-- (rate: 2400, 4800, 7200, 9600(default), 12000, 14400) -->
<!-- (ecm: 1:Enable(default), 0:Disable) -->
<!-- For GXW4104 (uncomment the following line by removing #) -->
<!-- P3602 = ch1-4:mode=1,rate=9600,ecm=1; -->
<!-- For GXW4108 (uncomment the following line by removing #) -->
<!-- P3602 = ch1-8:mode=1,rate=9600,ecm=1; -->
<!-- Channels Page -->
<!-- Phone Number Settings -->
<!-- FXO Port 1 -->
<!-- Channel(s), SIP USER ID, Authenticate ID, Password, Profile ID (3000 - Profile 1, 3001 - Profile 2, 3003 - Profile 3) -->
<P3030></P3030>
<P3060>{$user_id_1}</P3060>
<P3090>{$user_id_1}</P3090>
<P3120>{$user_password_1}</P3120>
<P3150>3000</P3150>
<!-- FXO Port 2 -->
<!-- Channel(s), SIP USER ID, Authenticate ID, Password, Profile ID (3000 - Profile 1, 3001 - Profile 2, 3003 - Profile 3) -->
<P3031></P3031>
<P3061>{$user_id_2}</P3061>
<P3091>{$user_id_2}</P3091>
<P3121>{$user_password_2}</P3121>
<P3151>3000</P3151>
<!-- FXO Port 3 -->
<!-- Channel(s), SIP USER ID, Authenticate ID, Password, Profile ID (3000 - Profile 1, 3001 - Profile 2, 3003 - Profile 3) -->
<P3032></P3032>
<P3062>{$user_id_3}</P3062>
<P3092>{$user_id_3}</P3092>
<P3122>{$user_password_3}</P3122>
<P3152>3000</P3152>
<!-- FXO Port 4 -->
<!-- Channel(s), SIP USER ID, Authenticate ID, Password, Profile ID (3000 - Profile 1, 3001 - Profile 2, 3003 - Profile 3) -->
<P3033></P3033>
<P3063>{$user_id_4}</P3063>
<P3093>{$user_id_4}</P3093>
<P3123>{$user_password_4}</P3123>
<P3153>3000</P3153>
<!-- FXO Port 5 (Uncomment the following P-values if using GXW4108) -->
<!-- Channel(s), SIP USER ID, Authenticate ID, Password, Profile ID (3000 - Profile 1, 3001 - Profile 2, 3003 - Profile 3) -->
<!-- P3034 = -->
<!-- P3064 = -->
<!-- P3094 = -->
<!-- P3124 = -->
<!-- P3154 = -->
<!-- FXO Port 6 (Uncomment the following P-values if using GXW4108) -->
<!-- Channel(s), SIP USER ID, Authenticate ID, Password, Profile ID (3000 - Profile 1, 3001 - Profile 2, 3003 - Profile 3) -->
<!-- P3035 = -->
<!-- P3065 = -->
<!-- P3095 = -->
<!-- P3125 = -->
<!-- P3155 = -->
<!-- FXO Port 7 (Uncomment the following P-values if using GXW4108) -->
<!-- Channel(s), SIP USER ID, Authenticate ID, Password, Profile ID (3000 - Profile 1, 3001 - Profile 2, 3003 - Profile 3) -->
<!-- P3036 = -->
<!-- P3066 = -->
<!-- P3096 = -->
<!-- P3126 = -->
<!-- P3156 = -->
<!-- FXO Port 8 (Uncomment the following P-values if using GXW4108) -->
<!-- Channel(s), SIP USER ID, Authenticate ID, Password, Profile ID (3000 - Profile 1, 3001 - Profile 2, 3003 - Profile 3) -->
<!-- P3037 = -->
<!-- P3067 = -->
<!-- P3097 = -->
<!-- P3127 = -->
<!-- P3157 = -->
<!-- Call Progress Tones -->
<!-- Dial Tone -->
<!-- For GXW4104 (uncomment the following line by removing #) -->
<!-- P3000 = ch1-4:f1=350@-11,f2=440@-11,c=0/0; -->
<!-- For GXW4108 (uncomment the following line by removing #) -->
<!-- P3000 = ch1-8:f1=350@-11,f2=440@-11,c=0/0; -->
<!-- Ringback Tone -->
<!-- For GXW4104 (uncomment the following line by removing #) -->
<!-- P3001 = ch1-4:f1=440@-11,f2=480@-11,c=200/400; -->
<!-- For GXW4108 (uncomment the following line by removing #) -->
<!-- P3001 = ch1-8:f1=440@-11,f2=480@-11,c=200/400; -->
<!-- Busy Tone -->
<!-- For GXW4104 (uncomment the following line by removing #) -->
<!-- P3002 = ch1-4:f1=480@-11,f2=620@-11,c=50/50; -->
<!-- For GXW4108 (uncomment the following line by removing #) -->
<!-- P3002 = ch1-8:f1=480@-11,f2=620@-11,c=50/50; -->
<!-- Reorder Tone -->
<!-- For GXW4104 (uncomment the following line by removing #) -->
<!-- P3003 = ch1-4:f1=480@-11,f2=620@-11,c=25/25; -->
<!-- For GXW4108 (uncomment the following line by removing #) -->
<!-- P3003 = ch1-8:f1=480@-11,f2=620@-11,c=25/25; -->
<!-- Channel Voice Settings -->
<!-- Tx to PSTN Audio Gain (dB). Default 1. Allowed values: -12-12. -->
<!-- For GXW4104 (uncomment the following line by removing #) -->
<!-- P3400 = ch1-4:1; -->
<!-- For GXW4108 (uncomment the following line by removing #) -->
<!-- P3400 = ch1-8:1; -->
<!-- Rx from PSTN Audio Gain (dB). Default 0. Allowed values: -12-12. -->
<!-- For GXW4104 (uncomment the following line by removing #) -->
<!-- P3401 = ch1-4:0; -->
<!-- For GXW4108 (uncomment the following line by removing #) -->
<!-- P3401 = ch1-8:0; -->
<!-- Silence Suppression. Default Yes. N - No, Y - Yes -->
<!-- For GXW4104 (uncomment the following line by removing #) -->
<!-- P3402 = ch1-4:Y; -->
<!-- For GXW4108 (uncomment the following line by removing #) -->
<!-- P3402 = ch1-8:Y; -->
<!-- Echo Cancellation. Default Yes. N - No, Y - Yes -->
<!-- For GXW4104 (uncomment the following line by removing #) -->
<!-- P3403 = ch1-4:Y; -->
<!-- For GXW4108 (uncomment the following line by removing #) -->
<!-- P3403 = ch1-8:Y; -->
<!-- Channel Specific Settings -->
<!-- DTMF Methods. Default 1. 1:in-audio, 2:RFC2833, 3:1+2, 4:SIP Info, 5:1+4, 6:2+4, 7:1+2+4 -->
<!-- For GXW4104 (uncomment the following line by removing #) -->
<!-- P3404 = ch1-4:1; -->
<!-- For GXW4108 (uncomment the following line by removing #) -->
<!-- P3404 = ch1-8:1; -->
<!-- No Key Entry Timeot. Allowed Values: 1-9, default 4. -->
<!-- For GXW4104 (uncomment the following line by removing #) -->
<!-- P3405 = ch1-4:4; -->
<!-- For GXW4108 (uncomment the following line by removing #) -->
<!-- P3405 = ch1-8:4; -->
<!-- Local SIP Listen Port -->
<!-- For GXW4104 (uncomment the following line by removing #) -->
<!-- P3406 = ch1-4:5060++; -->
<!-- For GXW4108 (uncomment the following line by removing #) -->
<!-- P3406 = ch1-8:5060++; -->
<!-- SRTP Mode(1-3). Default 1. -->
<!-- 1:disabled, 2:enabled but not forced, 3:enabled and forced -->
<!-- For GXW4104 (uncomment the following line by removing #) -->
<!-- P3409 = ch1-4:1; -->
<!-- For GXW4108 (uncomment the following line by removing #) -->
<!-- P3409 = ch1-8:1; -->
<!-- Port Scheduling Schema (Voip->PSTN) -->
<!-- Round-robin and/or Flexible -->
<!-- Syntax: rr: port_group; [...]), Default: rr:1-4; round-robin of all ports -->
<!-- For GXW4104 (uncomment the following line by removing #) -->
<!-- P3407 = rr:1-4; -->
<!-- For GXW4108 (uncomment the following line by removing #) -->
<!-- P3407 = rr:1-8; -->
<!-- Prefix to Specify Port(1 stage dialing method) -->
<!-- Syntax: prefix # + ch# + dialing# will request the ch# per call) -->
<!-- Note that this code has to prefix dialplan number and prefix doesn't impact round-robin -->
<P3408></P3408>
<!-- Dial Plan -->
<!-- PSTN Outgoing calls. -->
<P3331>{x+)</P3331>
<!-- Hookflash Duration (X10ms). Default 60. -->
<!-- For GXW4104 (uncomment the following line by removing #) -->
<!-- P3308 = ch1-4:60; -->
<!-- For GXW4108 (uncomment the following line by removing #) -->
<!-- P3308 = ch1-4:60; -->
<!-- Use DTMF Parameter from RFC2833 or SIP Info . 0 -Yes, 1 - No. Default Yes. -->
<P3190>0</P3190>
<!-- DTMF Digit Length(X10ms). Default 10. -->
<!-- For GXW4104 (uncomment the following line by removing #) -->
<!-- P3300 = ch1-4:10; -->
<!-- For GXW4108 (uncomment the following line by removing #) -->
<!-- P3300 = ch1-8:10; -->
<!-- DTMF Digit Volume(dB). Default 11. -->
<!-- For GXW4104 (uncomment the following line by removing #) -->
<!-- P3301 = ch1-4:-11; -->
<!-- For GXW4108 (uncomment the following line by removing #) -->
<!-- P3301 = ch1-4:-11; -->
<!-- DTMF Dial Pause Between Each Digit(X10ms) -->
<!-- For GXW4104 (uncomment the following line by removing #) -->
<!-- P3310 = ch1-4; -->
<!-- For GXW4108 (uncomment the following line by removing #) -->
<!-- P3310 = ch1-8; -->
<!-- End User Settings. Please do not edit this section. -->
<!-- Web Access. 0 - HTTP, 1 - HTTPS -->
<!-- P900 = 0 -->
<!-- Web Port: HTTP default is 80 and HTTPS default is 443 -->
<!-- P901 = 80 -->
<!-- End User Password -->
<!-- P196 = 123 -->
<!-- DHCP support. 0 - yes, 1 - no -->
<!-- P8 = 0 -->
<!-- DHCP hostname, alphabet, max. length is 32 -->
<!-- P146 = -->
<!-- DHCP domain, alphabet, max. length is 32 -->
<!-- P147 = -->
<!-- DHCP vendor class ID, alphabet, max. length is 32 -->
<!-- P148 = Grandstream GXW-410x -->
<!-- PPPoE support. PPPoE user ID -->
<!-- P82 = -->
<!-- PPPoE password -->
<!-- P83 = -->
<!-- PPPoE service name, max. length is 64 alpabit -->
<!-- P269 = -->
<!-- Preferred DNS server, four field, octet digits -->
<!-- P92 = -->
<!-- P93 = -->
<!-- P94 = -->
<!-- P95 = -->
<!-- IP Address. Ignore if DHCP or PPPoE is used -->
<!-- P9 = 192 -->
<!-- P10 = 168 -->
<!-- P11 = 0 -->
<!-- P12 = 1 -->
<!-- Subnet mask. Ignore if DHCP or PPPoE is used -->
<!-- P13 = 255 -->
<!-- P14 = 255 -->
<!-- P15 = 255 -->
<!-- P16 = 0 -->
<!-- Default Router. Ignore if DHCP or PPPoE is used -->
<!-- P17 = 192 -->
<!-- P18 = 168 -->
<!-- P19 = 1 -->
<!-- P20 = 1 -->
<!-- DNS 1. Ignore if DHCP or PPPoE is used -->
<!-- P21 = 192 -->
<!-- P22 = 168 -->
<!-- P23 = 0 -->
<!-- P24 = 1 -->
<!-- DNS 2. Ignore if DHCP or PPPoE is used -->
<!-- P25 = 0 -->
<!-- P26 = 0 -->
<!-- P27 = 0 -->
<!-- P28 = 0 -->
<!-- End User Time settings -->
<!-- Time Zone. Offset in minutes to GMT -->
<P64>420</P64>
<!-- Allow DHCP Option 2 to override Time Zone setting. 0 - No, 1 - Yes. -->
<!-- When set to Yes(1), it will override the configured Time Zone setting if available. -->
<P143>0</P143>
<!-- Daylight Savings Time. 0 - no, 1 - yes -->
<P75>0</P75>
<!-- Optional Rule: -->
<!-- If Daylight Saving Time is selected (P75 = 1), optional rule will allow automatically time ajustment based on the configured rule -->
<!-- Maxlength = 33, default is North America or US Daylight Saving Time Schecule: value="3,2,7,2,0;11,1,7,2,0;60" -->
<P246>3,2,7,2,0;11,1,7,2,0;60</P246>
</config>
</gs_provision>

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