Phones - A30 TXT Configuration 1.2.5
TXT Configuration
This section describes the formatting and options available when creating TXT-based configuration files for provisioning Asterisk A30 phones.
The configuration elements provided in this section are subject to change between Asterisk A-series phone firmware releases.
A Complete TXT Configuration Example
<<VOIP CONFIG FILE>>Version:2.0000000000
<NET CONFIG MODULE>
WAN IP :192.168.1.179
WAN Subnet Mask :255.255.255.0
WAN Gateway :192.168.1.1
Domain Name :
Primary DNS :8.8.8.8
Secondary DNS :202.96.134.133
Enable DHCP :1
DHCP Auto DNS :1
DHCP Auto Time :1
Use Vendor Class ID:0
Vendor Class ID :Asterisk
Enable PPPoE :0
PPPoE User :user123
PPPoE Password :password
--WIFI Config-- :
WIFI Enable :0
<MM CONFIG MODULE>
G723 Bit Rate :1
ILBC Payload Type :97
ILBC Payload Len :20
AMR Payload Type :108
AMRWB Payload Type :109
Dtmf Payload Type :101
Opus Payload Type :107
Opus Sample Rate :0
VAD :0
H264 Payload Type :117
Resv Audio Band :0
RTP Initial Port :10000
RTP Port Quantity :1000
RTP Keep Alive :0
RTCP CNAME User :
RTCP CNAME Host :
Select Your Tone :11
Sidetone GAIN :1
Dial Tone :350+440/0
Ringback Tone :440+480/2000,0/4000
Busy Tone :480+620/500,0/500
Congestion Tone :
Call waiting Tone :440/300,0/10000,440/300,0/10000,0/0
Holding Tone :
Error Tone :
Stutter Tone :
Information Tone :
Dial Recall Tone :350+440/100,0/100,350+440/100,0/100,350+440/100,0/100,350+440/0
Message Tone :
Howler Tone :
Number Unobtainable:400/500,0/6000
Warning Tone :1400/500,0/0
Record Tone :440/500,0/5000
Auto Answer Tone :
--PHONE CONFIG-- :
Audio Codec Sets :PCMU,PCMA,G726-32,G729,G723,iLBC,AMR,opus,G722
Video Codec Sets :H264
Video Frame Rate :25
Video Bit Rate :2000000
Video Resolution :4
<SIP CONFIG MODULE>
SIP Port :5060
STUN Server :
STUN Port :3478
STUN Refresh Time :50
SIP Wait Stun Time :800
Extern NAT Addrs :
Reg Fail Interval :32
SIP Pswd Encryption:0
Strict BranchPrefix:0
Video Mute Attr :0
Enable Group Backup:0
Enable RFC4475 :1
Strict UA Match :1
CSTA Enable :0
Notify Reboot :0
--SIP Line List-- :
SIP1 Phone Number :
SIP1 Display Name :
SIP1 Sip Name :
SIP1 Register Addr :
SIP1 Register Port :5060
SIP1 Register User :
SIP1 Register Pswd :
SIP1 Register TTL :3600
SIP1 Enable Reg :0
SIP1 Proxy Addr :
SIP1 Proxy Port :5060
SIP1 Proxy User :
SIP1 Proxy Pswd :
SIP1 BakProxy Addr :
SIP1 BakProxy Port :5060
SIP1 Enable Failback :0
SIP1 Signal Crypto :0
SIP1 SigCrypto Key :
SIP1 Media Crypto :0
SIP1 MedCrypto Key :
SIP1 SRTP Auth-Tag :0
SIP1 Local Domain :
SIP1 Always FWD :0
SIP1 Busy FWD :0
SIP1 No Answer FWD :0
SIP1 Always FWD Num :
SIP1 Busy FWD Num :
SIP1 NoAnswer FWD Num :
SIP1 FWD Timer :5
SIP1 Hotline Num :
SIP1 Enable Hotline :0
SIP1 WarmLine Time :0
SIP1 Pickup Num :
SIP1 Join Num :
SIP1 Intercom Num :
SIP1 Ring Type :default
SIP1 NAT UDPUpdate :2
SIP1 UDPUpdate TTL :60
SIP1 Server Type :0
SIP1 User Agent :
SIP1 PRACK :0
SIP1 Keep AUTH :0
SIP1 Session Timer :0
SIP1 S Timer Expires :0
SIP1 Enable GRUU :0
SIP1 DTMF Mode :3
SIP1 DTMF Info Mode :0
SIP1 NAT Type :0
SIP1 Enable Rport :1
SIP1 Subscribe :0
SIP1 Sub Expire :3600
SIP1 Single Codec :0
SIP1 CLIR :0
SIP1 Strict Proxy :1
SIP1 Direct Contact :0
SIP1 History Info :0
SIP1 DNS SRV :0
SIP1 DNS Mode :0
SIP1 XFER Expire :0
SIP1 Ban Anonymous :0
SIP1 Dial Off Line :0
SIP1 Quota Name :0
SIP1 Presence Mode :0
SIP1 RFC Ver :1
SIP1 Phone Port :0
SIP1 Signal Port :5060
SIP1 Transport :0
SIP1 Use SRV Mixer :0
SIP1 SRV Mixer Uri :
SIP1 Long Contact :0
SIP1 Auto TCP :0
SIP1 Uri Escaped :1
SIP1 Click to Talk :0
SIP1 MWI Num :
SIP1 CallPark Num :
SIP1 Retrieve Num :
SIP1 MSRPHelp Num :
SIP1 User Is Phone :0
SIP1 Auto Answer :0
SIP1 NoAnswerTime :5
SIP1 MissedCallLog :1
SIP1 SvcCode Mode :0
SIP1 DNDOn SvcCode :
SIP1 DNDOff SvcCode :
SIP1 CFUOn SvcCode :
SIP1 CFUOff SvcCode :
SIP1 CFBOn SvcCode :
SIP1 CFBOff SvcCode :
SIP1 CFNOn SvcCode :
SIP1 CFNOff SvcCode :
SIP1 ANCOn SvcCode :
SIP1 ANCOff SvcCode :
SIP1 Send ANOn Code :
SIP1 Send ANOffCode :
SIP1 CW On Code :
SIP1 CW Off Code :
SIP1 VoiceCodecMap :PCMU,PCMA,G726-32,G729,G723,iLBC,AMR,G722,AMR-WB
SIP1 VideoCodecMap :
SIP1 BLFList Uri :
SIP1 BLF Server :
SIP1 Respond 182 :0
SIP1 Enable BLFList :0
SIP1 Caller Id Type :4
SIP1 Syn Clock Time :0
SIP1 Use VPN :1
SIP1 Enable DND :0
SIP1 Inactive Hold :0
SIP1 Req With Port :0
SIP1 Update Reg Expire :1
SIP1 Enable SCA :0
SIP1 Sub CallPark :0
SIP1 Sub CC Status :0
SIP1 Feature Sync :0
SIP1 Enable XferBack :0
SIP1 XferBack Time :35
SIP1 Use Tel Call :0
SIP1 Enable Preview :0
SIP1 Preview Mode :1
SIP1 TLS Version :0
SIP1 CSTA Number :
SIP1 Enable ChgPort :0
SIP1 VQ Name :
SIP1 VQ Server :
SIP1 VQ Server Port :5060
SIP1 VQ HTTP Server :
SIP1 Flash Mode :0
SIP1 Content Type :
SIP1 Content Body :
SIP1 Unregister On Boot :0
SIP1 Enable MAC Header :0
SIP1 Record Start :Record:on
SIP1 Record Stop :Record:off
<CALL FEATURE MODULE>
--Port Config-- :
P1 Enable XferDPlan :1
P1 Enable FwdDPlan :0
P1 Enable Pre DPlan :0
P1 IP Dial Prefix :.
P1 Enable DND :1
P1 DND Mode :0
P1 Enable Space DND :0
P1 DND Start Time :1500
P1 DND End Time :1730
P1 Enable White List :1
P1 Enable Black List :1
P1 Enable CallBar :1
P1 Mute Ringing :0
P1 Ban Dial Out :0
P1 Ban Empty CID :0
P1 Accept Any Call :1
P1 Enable CLIP :1
P1 CallWaiting :1
P1 CallTransfer :1
P1 CallSemiXfer :1
P1 CallConference :1
P1 Auto PickupNext :0
P1 Busy No Line :1
P1 Auto Onhook :1
P1 Auto Onhook Time :3
P1 Enable Intercom :1
P1 Intercom Mute :0
P1 Intercom Tone :1
P1 Intercom Barge :1
P1 Use Auto Redial :0
P1 Redial EnterCallLog:0
P1 AutoRedial Delay :30
P1 AutoRedial Times :5
P1 Call Complete :0
P1 CHolding Tone :1
P1 CWaiting Tone :1
P1 Hide DTMF Type :0
P1 Talk DTMF Tone :1
P1 Dial DTMF Tone :1
P1 Psw Dial Mode :0
P1 Psw Dial Length :0
P1 Psw Dial Prefix :
P1 Enable MultiLine :1
P1 Allow IP Call :1
P1 Caller Name Type :0
P1 Mute For Ring :0
P1 Auto Handle Video :1
P1 Default Ans Mode :2
P1 Default Dial Mode :1
P1 Hold To Transfer :0
P1 Enable PreDial :1
P1 Default Ext Line :1
P1 Enable Def Line :0
P1 Enable SelLine :1
P1 Ring in Headset :0
P1 Auto Headset :0
P1 DND Return Code :480
P1 Busy Return Code :486
P1 Reject Return Code :603
P1 Contact Type :0
P1 Enable Country Code:0
P1 Country Code :
P1 Call Area Code :
P1 Number Privacy :0
P1 Privacy Rule :
P1 Transf DTMF Code :
P1 Hold DTMF Code :
P1 Conf DTMF Code :
--Basic DialPlan-- :
Dial by Pound :1
BTransfer by Pound :0
Onhook to BXfer :0
Onhook to AXfer :0
Conf Onhook to Xfer:0
Dial Fixed Length :0
Fixed Length Nums :11
Dial by Timeout :1
Dial Timeout value :10
Enable E OneSixFour:0
--Alert Info Ring--:
Alert1 Text :
Alert1 Ring Type :Type 1
<PHONE FEATURE MODULE>
Menu Password :789
KeyLock Password :123
Fast Keylock Code :
Enable KeyLock :0
KeyLock Timeout :0
Emergency Call :110
Push XML IP :
SIP Number Plan :0
LDAP Search :0
Search Path :0
Caller Display T :0
CallLog DisplayType:0
Enable Recv SMS :1
Enable Call History:1
Line Display Format:$name@$protocol$instance
Enable MWI Tone :0
--Display Input-- :
LCD Title :Asterisk
LCD Constrast :5
Enable Energysaving:4
LCD Luminance Level:12
Backlight Off Time :45
Disable CHN IME :0
Phone Model :
Host Name :bcm911188sv
Default Language :en
Enable Greetings :0
--Power LED-- :
Power :0
MWI Or SMS :3
In Using :0
Ring :2
Hold :0
Mute :0
Missed Call :3
--Voice Volume-- :
Handset Vol :5
Handset Mic Vol :3
Headset Vol :5
Headset Mic Vol :3
Headset Ring Vol :5
HandFree Vol :5
HandFree Mic Vol :3
HandFree Ring Vol :5
Ring Type :Type 2
--DateTime Config--:
Enable SNTP :1
SNTP Server :0.digium.pool.ntp.org
Second SNTP Server :1.digium.pool.ntp.org
Time Zone :-32
Time Zone Name :UTC-8
SNTP Timeout :5000
DST Type :1
DST Location :3
DST Rule Mode :0
DST Min Offset :60
DST Start Mon :3
DST Start Week :5
DST Start Wday :0
DST Start Hour :2
DST End Mon :10
DST End Week :5
DST End Wday :0
DST End Hour :2
--DateTime Display--:
Enable TimeDisplay :0
Time Display Style :0
Date Display Style :0
Date Separator :0
--ScreenSaver Config-- :
Screen Saver Type :0
Screen Timeout :0
Enable ActivePeriod:0
Period One Start :0
Period One End Time:0
Period Two Start :0
Period Two End Time:0
Screen Saver App :
Sleep After Active :0
Sleep Timeout :0
--Softkey Config-- :
Desktop Softkey :history;contact;dnd;menu;
Talking Softkey :hold;xfer;conf;end;
Ringing Softkey :accept;none;forward;reject;
Alerting Softkey :end;none;none;none;
XAlerting Softkey :end;none;none;none;
Conference Softkey :hold;none;split;end;
Waiting Softkey :hold;xfer;conf;end;
Ending Softkey :repeat;none;none;end;
DialerPre Softkey :send;2aB;delete;exit;
DialerCall Softkey :repeat;2aB;delete;exit;
DialerXfer Softkey :repeat;2aB;delete;exit;
DialerCfwd Softkey :repeat;2aB;delete;exit;
Desktop Click :history;status;none;none;none;
Dailer Click :none;none;none;none;none;
Call Click :none;none;voldown;volup;none;
Desktop Long Press :status;none;none;none;reset;
Softkey Mode :0
DialerConf Softkey :contact;clogs;redial;video;cancel;
-- Agent Config-- :
Agent Username :
Agent Password :
Agent Number :
Agent Sipline :0
Agent Status :0
Agent Status Reason:
--BW Directory-- :
BWDir1 Title :
BWDir1 URL :
BWDir1 Username :
BWDir1 Password :
BWDir1 SipLine :0
--BW Calllogs-- :
BWCLog1 Title :
BWCLog1 URL :
BWCLog1 Username :
BWCLog1 Password :
BWCLog1 SipLine :0
--LDAP Config-- :
LDAP1 Title :
LDAP1 Server :
LDAP1 port :389
LDAP1 Base :
LDAP1 Use SSL :0
LDAP1 Version :3
LDAP1 Calling Line :1
LDAP1 In Call Search :0
LDAP1 Out Call Search :0
LDAP1 Authenticate :3
LDAP1 Username :
LDAP1 Password :
LDAP1 Tel Attr :telephoneNumber
LDAP1 Mobile Attr :mobile
LDAP1 Other Attr :other
LDAP1 Name Attr :cn sn ou
LDAP1 Sort Attr :cn
LDAP1 Displayname :cn
LDAP1 Number Filter :(|(telephoneNumber=%)(mobile=%)(other=%))
LDAP1 Name Filter :(|(cn=%)(sn=%))
LDAP1 Max Hits :50
--Xml PhoneBook-- :
XML-PBook1 Name :
XML-PBook1 Addr :
XML-PBook1 UserName :
XML-PBook1 PassWd :
XML-PBook1 Sipline :0
<DEVICE MANAGER MODULE>
Onhook Time :120
<CTI CONFIG MODULE>
Enabled Active Uri :1
Enabled Action Url :1
Active Uri IP :
Start Reboot Url :
Boot Completed Url :
IP Change Url :
Reg On Url :
Reg Off Url :
Reg Failed Url :
PhoneState Idle Url:
PhoneState Talking :
PhoneState Ringing :
DND On Url :
DND Off Url :
Always FWD On Url :
Always FWD Off Url :
Busy FWD On Url :
Busy FWD Off Url :
No Ans FWD On Url :
No Ans FWD Off Url :
Mute On Url :
Mute Off Url :
Incmoing Call Url :
Outgoing Call Url :
Call Active Url :
Call Stop Url :
Transfer Url :
Hold On Url :
Hold Off Url :
Held On Url :
Held Off Url :
Mute On Call Url :
Mute Off Call Url :
New Missed call Url:
New MWI Url :
New SMS Url :
--CTI AT Config-- :
At Enabled :0
At Server :
<MCAST CFG MODULE>
Priority :0
Enable Priority :0
--Mcast Addr-- :
MCAST1 Name :
MCAST1 Host :
MCAST1 Port :0
<MMI CONFIG MODULE>
Web Server Type :0
Web Port :80
Https Web Port :443
Remote Control :1
Enable MMI Filter :0
Web Authentication :0
Enable Telnet :0
Telnet Port :23
Telnet Prompt :
Logon Timeout :15
--MMI Account-- :
Account1 Name :admin
Account1 Password :789
Account1 Level :10
<TR069 CONFIG MODULE>
TR069 Tone :1
CPE SerialNumber :
ACS Server Type :1
Enable TR069 :0
ACS URL :0.0.0.0
ACS UserName :admin
ACS Password :admin
ACS Backup URL :0.0.0.0
ACS BackupUserName :
ACS BackupPassword :
CPE UserName :dps
CPE Password :dps
Periodix Interval :3600
TLS Version :0
Area Code :020
STUN Enable :0
STUN Server Addr :
STUN Server Port :3478
STUN Local Port :30000
<SIP Hotspot MODULE>
Enable Hotspot :0
Mode :1
Listen Type :0
Listen IP :224.0.2.0
Listen Port :16360
Own Name :SIP Hotspot
--Line Conf List-- :
HS1 Enable :1
<VPN CONFIG MODULE>
VPN mode :-1
Enable VPN :0
Enable Nat :0
Openvpn mode :0
L2TP Server Address:
L2TP User Name :
L2TP Password :
L2TP Negotiate DNS :1
PPTP Server Address:
PPTP User Name :
PPTP Password :
<MAINTENANCE CONFIG MODULE>
Contact Update Mode:0
<AUTOUPDATE CONFIG MODULE>
Default Username :
Default Password :
Input Cfg File Name:
Device Cfg File Key:
Common Cfg File Key:
Download CommonConf:1
Save Provision Info:0
Check FailTimes :5
Flash Server IP :
Flash File Name :
Flash Protocol :2
Flash Mode :0
Flash Interval :1
update PB Interval :720
--Sip Pnp List-- :
PNP Enable :1
PNP IP :224.0.1.75
PNP Port :5060
PNP Transport :0
PNP Interval :1
--Net Option-- :
DHCP Option :66
Dhcp Option 120 :0
<OTA CONFIG MODULE>
<RPS CONFIG MODULE>
Rps Name :digium
<DIGIUM CONFIG MODULE>
Digium Enable :1
Lockdown Status :0
Digium Url :https://phoneservice.digium.com/json
Pbx User Agent :
<FIRMWARE CHECK MODULE>
Enable Auto Upgrade:0
Upgrade Server 1 :
Upgrade Server 2 :
Auto Upgrade Interval:24
<QOS CONFIG MODULE>
Enable VLAN :0
VLAN ID :256
Enable PVID :0
PVID Value :254
Signalling Priority:0
Voice Priority :0
Video Priority :0
Enable diffServ :1
Singalling DSCP :46
Voice DSCP :46
Video DSCP :46
LLDP Transmit :1
LLDP Refresh Time :60
LLDP Learn Policy :1
LLDP Save Learn Data:0
CDP Enable :0
CDP Refresh Time :60
DHCP Option Vlan :0
<LOG CONFIG MODULE>
Level :INFO
Style :level,tag
Output Device :stdout
File Name :platform.log
File Size :512KB
Syslog Tag :platform
Syslog Server :0.0.0.0
Syslog Server Port :514
<APP CONFIG MODULE>
Watch Dog Enabled :1
Enable In Access :0
Enable Out Access :0
<VQM CONFIG MODULE>
Session Report :1
Interval Report :1
Interval Period :60
MOS-LQ Warning :40
MOS-LQ Critical :25
Delay Warning :150
Delay Critical :200
Phone Report :1
WEB Report :1
<DOT1X CONFIG MODULE>
Xsup Mode :0
Xsup User :admin
Xsup Password :admin
--SSL Mode-- :
Permission CTF :0
Common Name :0
CTF mode :0
<RECORD CONFIG MODULE>
Enabled :1
Voice Codec :G729
Record Type :0
File Size Limit :8
Server Addr :
Server Port :0
<ATE CONFIG MODULE>
ATE Id :0000000000000000
<UI MAINTAIN CONFIG MODULE>
Timeout To Screensaver :0
User Change Background :0
EHS Headset type :0
<DSSKEY CONFIG MODULE>
Select DsskeyAction:0
Memory Key to BXfer:3
FuncKey Page Num :5
DSS Home Page :0
Expand Board Enable:0
Extern1 Page Belong :0
--Dsskey Config1--:
Fkey1 Type :2
Fkey1 Value :SIP1
Fkey1 Title :
--SoftDss Config-- :
Fkey1 Type :0
Fkey1 Value :
Fkey1 Title :
<<END OF FILE>> Top Level Elements
Elements Example
<<VOIP CONFIG FILE>>Version:2.0000000000
<<END OF FILE>>Each element is populated with a value.
Top Level Elements
Option | Values | Description |
|---|---|---|
Version | String | If the phone is configured for auto provisioning and it successfully loads this configuration, the version string will be displayed on-screen. |
NET CONFIG MODULE Elements
Elements Example
<<VOIP CONFIG FILE>>
<NET CONFIG MODULE>
WAN IP :192.168.1.179
WAN Subnet Mask :255.255.255.0
WAN Gateway :192.168.1.1
Domain Name :
Primary DNS :8.8.8.8
Secondary DNS :202.96.134.133
Enable DHCP :1
DHCP Auto DNS :1
DHCP Auto Time :1
Use Vendor Class ID:0
Vendor Class ID :Asterisk
Enable PPPoE :0
PPPoE User :user123
PPPoE Password :password
--WIFI Config-- :
WIFI Enable :0
<<END OF FILE>>Each element is populated with a value.
NET CONFIG MODULE Elements
Option | Values | Description |
|---|---|---|
WAN IP | IPv4 address | Sets the IPv4 address of the phone. |
WAN Subnet Mask | IPv4 net mask | Sets the IPv4 net mask of the phone. |
WAN Gateway | IPv4 network gateway | Sets the IPv4 default gateway / route of the phone. |
Domain Name | Domain name, as string | Sets the default domain search name of the phone. |
Primary DNS | IPv4 address | Sets the primary DNS server of the phone. |
Secondary DNS | IPv4 address | Sets the secondary DNS server of the phone. |
Enable DHCP | Boolean, Defaults to 1 | Enables or disables DHCP network configuration request. Defaults to 1. |
DHCP Auto DNS | Boolean, Defaults to 1 | Configures the phone to accept DNS servers provided by DHCP. If disabled, the phone will use the DNS servers defined by the PrimaryDNS and SecondaryDNS net configuration elements. Defaults to 1. |
DHCP Auto Time | Boolean, Defaults to 1 | Configures the phone to set its time and SNTP server based upon receipt of DHCP Option 42. If disabled, the phone will use the NTP server as defined by the SNTPServer date configuration element and time as defined in other date configuration elements. Defaults to 1. |
Use Vendor Class ID | Boolean, Defaults to 0 | Configures the phone to send DHCP Option 61, Vendor Class Identifier. Defaults to 0. |
Enable PPPoE | Boolean, Defaults to 0 | Configures the phone to perform PPPoE authentication in order to retrieve its network configuration. Defaults to 0 |
PPPoE User | String | If the phone is to use PPPoE authentication, sets the PPPoE authentication username. |
PPPoE Password | String | If the phone is to use PPPoE authentication, sets the PPPoE authentication password. |
WIFI Config Element (currently unused)
This section contains child elements controlling WIFI support
Child elements of WIFI Config
Elements Example
<<VOIP CONFIG FILE>>
<NET CONFIG MODULE>
--WIFI Config-- :
WIFI Enable :0
<<END OF FILE>>Option | Values | Description |
|---|---|---|
WIFI Enable | Boolean, Defaults to 0 | Enables or disables support for WiFi network connectivity. Currently unused. |
MM CONFIG MODULE Elements
Elements
<<VOIP CONFIG FILE>>
<MM CONFIG MODULE>
G723 Bit Rate :1
ILBC Payload Type :97
ILBC Payload Len :20
AMR Payload Type :108
AMRWB Payload Type :109
Dtmf Payload Type :101
Opus Payload Type :107
Opus Sample Rate :0
VAD :0
H264 Payload Type :117
Resv Audio Band :0
RTP Initial Port :10000
RTP Port Quantity :1000
RTP Keep Alive :0
RTCP CNAME User :
RTCP CNAME Host :
Select Your Tone :11
Sidetone GAIN :1
Dial Tone :350+440/0
Ringback Tone :440+480/2000,0/4000
Busy Tone :480+620/500,0/500
Congestion Tone :
Call waiting Tone :440/300,0/10000,440/300,0/10000,0/0
Holding Tone :
Error Tone :
Stutter Tone :
Information Tone :
Dial Recall Tone :350+440/100,0/100,350+440/100,0/100,350+440/100,0/100,350+440/0
Message Tone :
Howler Tone :
Number Unobtainable:400/500,0/6000
Warning Tone :1400/500,0/0
Record Tone :440/500,0/5000
Auto Answer Tone :
--PHONE CONFIG-- :
Audio Codec Sets :PCMU,PCMA,G726-32,G729,G723,iLBC,AMR,opus,G722
Video Codec Sets :H264
Video Frame Rate :25
Video Bit Rate :2000000
Video Resolution :4
<<END OF FILE>>Each element is populated with a value.
MM CONFIG MODULE Elements
Option | Values | Description |
|---|---|---|
G723 Bit Rate | Boolean, Defaults to 1 | Sets the bitrate type to be used for G.723.1 voice codec encoding. 0 sets encoding to 5.3kbit/s. 1 sets encoding to 6.3kbit/s. Defaults to 1 |
ILBC Payload Type | Integer, Defaults to 97. | Sets the RTP payload type to be used for iLBC voice codec encoding. Defaults to 97. |
ILBC Payload Len | 20, 30 | Sets the iLBC voice codec block length duration in milliseconds. 20 sets 20ms. 30 sets 30ms. Defaults to 20. |
AMR Payload Type | Integer, Defaults to 108 | Sets the RTP payload type to be used for AMR voice codec encoding. Defaults to 108. |
AMRWB Payload Type | Integer, Defaults to 109 | Sets the RTP payload type to be used for AMR (G.722.2) voice codec encoding. Defaults to 109. |
Dtmf Payload Type | Integer, Defaults to 101 | Sets the RTP payload type to be used for RFC2833 DTMF events. Defaults to 101. |
Opus Payload Type | Integer, Defaults to 107 | Sets the RTP payload type to be used for Opus voice codec encoding. Defaults to 107 |
Opus Sample Rate | Boolean, Defaults to 0 | Sets the Opus voice codec encoding type. 0 sets Narrowband. 1 sets Wideband. Defaults to 0. |
VAD | Boolean, Defaults to 0 | Enables or disables Voice Activity Detection (VAD) for certain (G.729) codecs. Defaults to 0. |
H264 Payload Type | Integer, Defaults to 117 | Sets the RTP payload type to be used for H.264 video codec encoding. Defaults to 117 |
Resv Audio Band | Boolean, Defaults to 0 | If enabled, the phone will reduce video bandwidth to prioritize audio. Defaults to 0. |
RTP Initial Port | Integer, Defaults to 10000 | Sets the starting RTP port used for calls. Defaults to 10000 |
RTP Port Quantity | Integer, Defaults to 200 | Sets the number of RTP ports to span across before recycling. Defaults to 1000. |
RTP Keep Alive | Boolean, Defaults to 0 | Enables or disables the sending of a periodic RTP keep alive packet. Defaults to 0. |
RTCP CNAME User | String | Sets the RTCP CNAME User part for RTCP Sender reports generated by the phone. |
RTCP CNAME Host | String | Sets the RTCP CNAME Host part for RTCP Sender reports generated by the phone. |
Select Your Tone | Integer, Defaults to 11 | Sets the phone's default tones based upon country as follows: Australia, 15 |
Dial Tone | String | Sets the dialing tone pattern, as a tone or combination of tones with a defined duration, optionally followed by other tones, e.g. 350+440/0 |
Ringback Tone | String | Sets the ring back tone pattern, as a tone or combination of tones with a defined duration, optionally followed by other tones, e.g. |
Busy Tone | String | Sets the ring back tone pattern, as a tone or combination of tones with a defined duration, optionally followed by other tones, e.g. |
Congestion Tone | String | Sets the congestion tone pattern, as a tone or combination of tones with a defined duration, optionally followed by other tones. |
Call waiting Tone | String | Sets the call waiting tone pattern, as a tone or combination of tones with a defined duration, optionally followed by other tones, e.g. |
Holding Tone | String | Sets the holding tone pattern, as a tone or combination of tones with a defined duration, optionally followed by other tones. |
Error Tone | String | Sets the error tone pattern, as a tone or combination of tones with a defined duration, optionally followed by other tones. |
Stutter Tone | String | Sets the stutter tone pattern, as a tone or combination of tones with a defined duration, optionally followed by other tones. |
Information Tone | String | Sets the information tone pattern, as a tone or combination of tones with a defined duration, optionally followed by other tones. |
Dial Recall Tone | String | Sets the dial recall tone pattern, as a tone or combination of tones with a defined duration, optionally followed by other tones, e.g. 350+440/100,0/100,350+440/100,0/100,350+440/100,0/100,350+440/0 |
Message Tone | String | Sets the message tone pattern, as a tone or combination of tones with a defined duration, optionally followed by other tones. |
Howler Tone | String | Sets the howler tone pattern, as a tone or combination of tones with a defined duration, optionally followed by other tones. |
Number Unobtainable | String | Sets the number unobtainable tone pattern, as a tone or combination of tones with a defined duration, optionally followed by other tones, e.g. 400/500,0/6000 |
Warning Tone | String | Sets the warning tone pattern, as a tone or combination of tones with a defined duration, optionally followed by other tones, e.g. 1400/500,0/0 |
Record Tone | String | Sets the record tone pattern, as a tone or combination of tones with a defined duration, optionally followed by other tones, e.g. 440/500,0/5000 |
Auto Answer Tone | String | Sets the auto answer tone pattern, as a tone or combination of tones with a defined duration, optionally followed by other tones. |
PHONE CONFIG Element
This section contains child elements controlling audio and video codecs.
Child elements of PHONE CONFIG
Elements Example
<<VOIP CONFIG FILE>>
<MM CONFIG MODULE>
--PHONE CONFIG-- :
Audio Codec Sets :PCMU,PCMA,G726-32,G729,G723,iLBC,AMR,opus,G722
Video Codec Sets :H264
Video Frame Rate :25
Video Bit Rate :2000000
Video Resolution :4
<<END OF FILE>>Option | Values | Description |
|---|---|---|
Audio Codec Sets | Comma-separated list of: PCMU | Sets the list of codecs the phone is allowed to transcode. |
Video Codec Sets | H264 | Sets the video codec the A30 is allowed to decode. |
Video Frame Rate | Integer, Defaults to 25 | Sets the frame rate the A30 should expect to decode. |
Video Bit Rate | Integer, Defaults to 2000000 | Sets the bitrate of the video the A30 should expect to decode. |
Video Resolution | Integer, Defaults to 4 | Sets the resolution of the video the A30 should expect to decode. |
SIP CONFIG MODULE Elements
Elements
<<VOIP CONFIG FILE>>
<SIP CONFIG MODULE>
SIP Port :5060
STUN Server :
STUN Port :3478
STUN Refresh Time :50
SIP Wait Stun Time :800
Extern NAT Addrs :
Reg Fail Interval :32
SIP Pswd Encryption:0
Strict BranchPrefix:0
Video Mute Attr :0
Enable Group Backup:0
Enable RFC4475 :1
Strict UA Match :1
CSTA Enable :0
Notify Reboot :0
--SIP Line List-- :
SIP1 Phone Number :
SIP1 Display Name :
SIP1 Sip Name :
SIP1 Register Addr :
SIP1 Register Port :5060
SIP1 Register User :
SIP1 Register Pswd :
SIP1 Register TTL :3600
SIP1 Enable Reg :0
SIP1 Proxy Addr :
SIP1 Proxy Port :5060
SIP1 Proxy User :
SIP1 Proxy Pswd :
SIP1 BakProxy Addr :
SIP1 BakProxy Port :5060
SIP1 Enable Failback :0
SIP1 Signal Crypto :0
SIP1 SigCrypto Key :
SIP1 Media Crypto :0
SIP1 MedCrypto Key :
SIP1 SRTP Auth-Tag :0
SIP1 Local Domain :
SIP1 Always FWD :0
SIP1 Busy FWD :0
SIP1 No Answer FWD :0
SIP1 Always FWD Num :
SIP1 Busy FWD Num :
SIP1 NoAnswer FWD Num :
SIP1 FWD Timer :5
SIP1 Hotline Num :
SIP1 Enable Hotline :0
SIP1 WarmLine Time :0
SIP1 Pickup Num :
SIP1 Join Num :
SIP1 Intercom Num :
SIP1 Ring Type :default
SIP1 NAT UDPUpdate :2
SIP1 UDPUpdate TTL :60
SIP1 Server Type :0
SIP1 User Agent :
SIP1 PRACK :0
SIP1 Keep AUTH :0
SIP1 Session Timer :0
SIP1 S Timer Expires :0
SIP1 Enable GRUU :0
SIP1 DTMF Mode :3
SIP1 DTMF Info Mode :0
SIP1 NAT Type :0
SIP1 Enable Rport :1
SIP1 Subscribe :0
SIP1 Sub Expire :3600
SIP1 Single Codec :0
SIP1 CLIR :0
SIP1 Strict Proxy :1
SIP1 Direct Contact :0
SIP1 History Info :0
SIP1 DNS SRV :0
SIP1 DNS Mode :0
SIP1 XFER Expire :0
SIP1 Ban Anonymous :0
SIP1 Dial Off Line :0
SIP1 Quota Name :0
SIP1 Presence Mode :0
SIP1 RFC Ver :1
SIP1 Phone Port :0
SIP1 Signal Port :5060
SIP1 Transport :0
SIP1 Use SRV Mixer :0
SIP1 SRV Mixer Uri :
SIP1 Long Contact :0
SIP1 Auto TCP :0
SIP1 Uri Escaped :1
SIP1 Click to Talk :0
SIP1 MWI Num :
SIP1 CallPark Num :
SIP1 Retrieve Num :
SIP1 MSRPHelp Num :
SIP1 User Is Phone :0
SIP1 Auto Answer :0
SIP1 NoAnswerTime :5
SIP1 MissedCallLog :1
SIP1 SvcCode Mode :0
SIP1 DNDOn SvcCode :
SIP1 DNDOff SvcCode :
SIP1 CFUOn SvcCode :
SIP1 CFUOff SvcCode :
SIP1 CFBOn SvcCode :
SIP1 CFBOff SvcCode :
SIP1 CFNOn SvcCode :
SIP1 CFNOff SvcCode :
SIP1 ANCOn SvcCode :
SIP1 ANCOff SvcCode :
SIP1 Send ANOn Code :
SIP1 Send ANOffCode :
SIP1 CW On Code :
SIP1 CW Off Code :
SIP1 VoiceCodecMap :PCMU,PCMA,G726-32,G729,G723,iLBC,AMR,G722,AMR-WB
SIP1 VideoCodecMap :
SIP1 BLFList Uri :
SIP1 BLF Server :
SIP1 Respond 182 :0
SIP1 Enable BLFList :0
SIP1 Caller Id Type :4
SIP1 Syn Clock Time :0
SIP1 Use VPN :1
SIP1 Enable DND :0
SIP1 Inactive Hold :0
SIP1 Req With Port :0
SIP1 Update Reg Expire :1
SIP1 Enable SCA :0
SIP1 Sub CallPark :0
SIP1 Sub CC Status :0
SIP1 Feature Sync :0
SIP1 Enable XferBack :0
SIP1 XferBack Time :35
SIP1 Use Tel Call :0
SIP1 Enable Preview :0
SIP1 Preview Mode :1
SIP1 TLS Version :0
SIP1 CSTA Number :
SIP1 Enable ChgPort :0
SIP1 VQ Name :
SIP1 VQ Server :
SIP1 VQ Server Port :5060
SIP1 VQ HTTP Server :
SIP1 Flash Mode :0
SIP1 Content Type :
SIP1 Content Body :
SIP1 Unregister On Boot :0
SIP1 Enable MAC Header :0
SIP1 Record Start :Record:on
SIP1 Record Stop :Record:off
<MM CONFIG MODULE>
--PHONE CONFIG-- :
Audio Codec Sets :PCMU,PCMA,G726-32,G729,G723,iLBC,AMR,opus,G722
Video Codec Sets :H264
Video Frame Rate :25
Video Bit Rate :2000000
Video Resolution :4
<<END OF FILE>>Each element is populated with a value.
SIP CONFIG MODULE Elements
Option | Values | Description |
|---|---|---|
SIP Port | Integer, 1-65535 | Sets the local SIP signaling port used by the phone. Defaults to 5060 |
STUN Server | IPv4 address or hostname | Sets the IPv4 address or hostname of a remote STUN server to be used by the phone. Defaults to none. |
STUN Port | Integer, 1-65535 | Sets the port of the remote STUN server to be used by the phone. Defaults to none |
STUN Refresh Time | Integer, in seconds | Sets the STUN server refresh period used by the phone. Defaults to 50. |
SIP Wait Stun Time | Integer, in milliseconds | Sets the amount of the, in milliseconds, the phone should wait for the STUN server to respond before continuing. |
Extern NAT Addrs | IPv4 address | If the phone is configured to use STUN, the SIP Contact address will be set according to this option. IP address of the phone, as broadcast by an upstream NAT router. |
Reg Fail Interval | Integer, in seconds | Sets the time, after a failed registration, after reaching the halfway point of this interval time, at which the phone will attempt registration again. Defaults to 32. |
SIP Pswd Encryption | Boolean, Defaults to 0 | If enabled, passwords contained in exported phone configurations will be disguised. Defaults to 0. |
Strict BranchPrefix | Boolean, Defaults to 0 | If enabled, when the phone receives a SIP message with a branch field that does not begin with z9hG4bk, the phone will return a 4xx message. Defaults to 0 |
Video Mute Attr | Boolean, Defaults to 0 | If enabled, causes the phone, when placing a video-enabled call on hold, to use the "inactive" SDP a-line parameter. If disabled, the phone will use the sendrecv parameter instead. Defaults to 0 |
Enable Group Backup | Boolean, Defaults to 0 | If enabled, and the phone has both regular and backup registrars configured, the phone will, upon a failure to register with either of the registrars, unregister from both. If disabled, the phone will attempt to register with both, and the registration of one will not affect the registration with another. Defaults to 0. |
Enable RFC4475 | Boolean, Defaults to 0 | If enabled, and the phone receives a SIP message with a From or To field containing a blank space, quotation marks, or both, the phone will reject the message. Defaults to 0. |
Strict UA Match | Boolean, Defaults to 0 | If enabled, the phone will only respond to requests from servers to which it is registered, based on the user-agent string provided by the incoming request. Defaults to 0. |
CSTA Enable | Boolean, Defaults to 0 | Enables or disables the phone's support of uaCSTA. Defaults to 0. |
Notify Reboot | Boolean, Defaults to 0 | If enabled, the phone will reboot in response to a received check-sync Event. |
SIP Line List Element
This section contains child elements controlling SIP Line configurations. Each line is controlled by its index value, beginning with 1, up to 6.
Child elements of SIP Line List
Elements Example
<<VOIP CONFIG FILE>>
<SIP CONFIG MODULE>
--SIP Line List-- :
SIP1 Phone Number :
SIP1 Display Name :
SIP1 Sip Name :
SIP1 Register Addr :
SIP1 Register Port :5060
SIP1 Register User :
SIP1 Register Pswd :
SIP1 Register TTL :3600
SIP1 Enable Reg :0
SIP1 Proxy Addr :
SIP1 Proxy Port :5060
SIP1 Proxy User :
SIP1 Proxy Pswd :
SIP1 BakProxy Addr :
SIP1 BakProxy Port :5060
SIP1 Enable Failback :0
SIP1 Signal Crypto :0
SIP1 SigCrypto Key :
SIP1 Media Crypto :0
SIP1 MedCrypto Key :
SIP1 SRTP Auth-Tag :0
SIP1 Local Domain :
SIP1 Always FWD :0
SIP1 Busy FWD :0
SIP1 No Answer FWD :0
SIP1 Always FWD Num :
SIP1 Busy FWD Num :
SIP1 NoAnswer FWD Num :
SIP1 FWD Timer :5
SIP1 Hotline Num :
SIP1 Enable Hotline :0
SIP1 WarmLine Time :0
SIP1 Pickup Num :
SIP1 Join Num :
SIP1 Intercom Num :
SIP1 Ring Type :default
SIP1 NAT UDPUpdate :2
SIP1 UDPUpdate TTL :60
SIP1 Server Type :0
SIP1 User Agent :
SIP1 PRACK :0
SIP1 Keep AUTH :0
SIP1 Session Timer :0
SIP1 S Timer Expires :0
SIP1 Enable GRUU :0
SIP1 DTMF Mode :3
SIP1 DTMF Info Mode :0
SIP1 NAT Type :0
SIP1 Enable Rport :1
SIP1 Subscribe :0
SIP1 Sub Expire :3600
SIP1 Single Codec :0
SIP1 CLIR :0
SIP1 Strict Proxy :1
SIP1 Direct Contact :0
SIP1 History Info :0
SIP1 DNS SRV :0
SIP1 DNS Mode :0
SIP1 XFER Expire :0
SIP1 Ban Anonymous :0
SIP1 Dial Off Line :0
SIP1 Quota Name :0
SIP1 Presence Mode :0
SIP1 RFC Ver :1
SIP1 Phone Port :0
SIP1 Signal Port :5060
SIP1 Transport :0
SIP1 Use SRV Mixer :0
SIP1 SRV Mixer Uri :
SIP1 Long Contact :0
SIP1 Auto TCP :0
SIP1 Uri Escaped :1
SIP1 Click to Talk :0
SIP1 MWI Num :
SIP1 CallPark Num :
SIP1 Retrieve Num :
SIP1 MSRPHelp Num :
SIP1 User Is Phone :0
SIP1 Auto Answer :0
SIP1 NoAnswerTime :5
SIP1 MissedCallLog :1
SIP1 SvcCode Mode :0
SIP1 DNDOn SvcCode :
SIP1 DNDOff SvcCode :
SIP1 CFUOn SvcCode :
SIP1 CFUOff SvcCode :
SIP1 CFBOn SvcCode :
SIP1 CFBOff SvcCode :
SIP1 CFNOn SvcCode :
SIP1 CFNOff SvcCode :
SIP1 ANCOn SvcCode :
SIP1 ANCOff SvcCode :
SIP1 Send ANOn Code :
SIP1 Send ANOffCode :
SIP1 CW On Code :
SIP1 CW Off Code :
SIP1 VoiceCodecMap :PCMU,PCMA,G726-32,G729,G723,iLBC,AMR,G722,AMR-WB
SIP1 VideoCodecMap :
SIP1 BLFList Uri :
SIP1 BLF Server :
SIP1 Respond 182 :0
SIP1 Enable BLFList :0
SIP1 Caller Id Type :4
SIP1 Syn Clock Time :0
SIP1 Use VPN :1
SIP1 Enable DND :0
SIP1 Inactive Hold :0
SIP1 Req With Port :0
SIP1 Update Reg Expire :1
SIP1 Enable SCA :0
SIP1 Sub CallPark :0
SIP1 Sub CC Status :0
SIP1 Feature Sync :0
SIP1 Enable XferBack :0
SIP1 XferBack Time :35
SIP1 Use Tel Call :0
SIP1 Enable Preview :0
SIP1 Preview Mode :1
SIP1 TLS Version :0
SIP1 CSTA Number :
SIP1 Enable ChgPort :0
SIP1 VQ Name :
SIP1 VQ Server :
SIP1 VQ Server Port :5060
SIP1 VQ HTTP Server :
SIP1 Flash Mode :0
SIP1 Content Type :
SIP1 Content Body :
SIP1 Unregister On Boot :0
SIP1 Enable MAC Header :0
SIP1 Record Start :Record:on
SIP1 Record Stop :Record:off
<<END OF FILE>>Option | Values | Description |
|---|---|---|
Phone Number | String | Sets the identifier used for the user part of the From and To lines in the phone's SIP messaging. |
Display Name | String | Sets the quoted string name identifier used in the From and To (REGISTER only) in the phone's SIP messaging. |
Sip Name | String | Sets a name for the SIP line, visible within the web admin UI of the phone. |
Register Addr | IPv4 address or hostname | Sets the IPv4 address or hostname of the SIP Registrar. |
Register Port | Integer, 1-65535 | Sets the port used to contact the SIP Registrar. Defaults to 5060 |
Register User | String | Sets the SIP authentication user. |
Register Pswd | String | Sets the SIP authentication password. |
Register TTL | Integer, in seconds, Defaults to 3600 | Sets the default Expires timer for SIP registration. Defaults to 3600. |
Enable Reg | Boolean, Defaults to 0 | Enables or disables SIP registration for this line. |
Proxy Addr | IPv4 address or hostname | Sets the IPv4 address or hostname for the SIP Proxy. Defaults to none. |
Proxy Port | Integer, 1-65535 | Sets the port used to contact the SIP Proxy. Defaults to 5060. |
Proxy User | String | Sets the SIP proxy authentication user. |
Proxy Pswd | String | Sets the SIP proxy authentication password. |
BakProxy Addr | IPv4 address or hostname | Sets the IPv4 address or hostname for the backup SIP Proxy. Defaults to none. |
BakProxy Port | Integer, 1-65535 | Sets the port used to contact the backup SIP Proxy. Defaults to 5060. |
Enable Failback | Boolean, Defaults to 0 | Sets whether the phone, whereby it is configured with both a primary and a backup proxy, should, whereupon detection that the primary proxy is operational again after having previously failed, resume sending calls to the primary server. If enabled, the phone will switch back to the primary proxy. If disabled, the phone will continue to send calls to the backup. Defaults to 0. |
Signal Crypto | Boolean, Defaults to 0 | If enabled, causes the phone to encrypt SIP signaling. Defaults to 0. |
SigCrypto Key | String | Sets the key used to encrypt SIP signaling. |
Media Crypto | Boolean, Defaults to 0 | If enabled, causes the phone to encrypt RTP media. Defaults to 0. |
MedCrypto Key | String | Sets the key used to encrypt RTP media. |
SRTP Auth-Tag | Intege, Defaults to 0 | If set to 0, the phone will utilizes 80 byte SRTP tags, encrypting RTP using AES_CM_128_HMAC_SHA_80. If set to 1, the phone will utilize 32 byte SRTP tags, encrypting RTP using AES_CM_128_HMAC_SHA_32. Defaults to 0. |
Local Domain | Strings | Sets the domain name used in SIP registration |
Always FWD | Boolean, Defaults to 0 | If enabled, the phone will unconditionally forward calls. Defaults to 0. |
Busy FWD | Boolean, Defaults to 0 | If enabled, the phone will forward calls whenever it is busy. Defaults to 0. |
No Answer FWD | Boolean, Defaults to 0 | If enabled, the phone will forward calls that are not answered. Defaults to 0. |
Always FWD Num | String | Sets the forwarding number used in conjunction with the AlwaysFWD option. |
Busy FWD Num | String | Sets the forwarding number used in conjunction with the BusyFWD option. |
NoAnswer FWD Num | String | Sets the forwarding number used in conjunction with the NoAnswerFWD option. |
FWD Timer | Integer, in seconds, Defaults to 5 | Sets the time, in seconds, applied to the NoAnswerFWD option. Defaults to 5. |
Hotline Num | String | Sets the number to be dialed when the off-hook time is greater than the WarmLineTime and EnableHotline is enabled. |
Enable Hotline | Boolean, Defaults to 0 | If enabled, and the phone has been off-hook longer than the WarmLineTime, the phone will automatically dial the number defined by HotlineNum. |
WarmLine Time | Integer, in seconds, 0-9, Defaults to 0. | Sets the amount of time the phone must remain off-hook before attempting to execute Hotline functionality. |
Pickup Num | String | Sets the dialing prefix applied to calls that are picked up using function keys configured for Pickup functionality. |
Join Num | String | Sets the dialing prefix applied to calls that are joined using function keys configured for Join functionality. |
Ring Type | Integer, 1-9, Defaults to 1 | Sets the default ringing type to be used. Defaults to 1. |
NAT UDPUpdate | Integer, 0-2, Defaults to 1 | If set to 1, the phone will send SIP OPTION packets to the server after each UDPUpdate_TTL time. If set to 2, the phone will send a CRLF to the server after each UDPUpdate_TTL time. If set to 0, the phone will send nothing. Defaults to 1. |
UDPUpdate TTL | Integer, in seconds, Defaults to 60 | Sets the timer used by the NATUDPUpdate option. |
Server Type | Integer | Sets special compatibility settings required for specific server types. The following types are supported: 3CX - 31 Defaults to 0. |
User Agent | String | Sets the SIP User-Agent passed by the phone when communicating. SHOULD THIS DEFAULT TO A30? |
PRACK | Boolean, Defaults to 0 | Enables or disables SIP PRACK functionality within the phone. Defaults to 0. |
Keep AUTH | Boolean, Defaults to 0 | If enabled, the phone will, on SIP re-registration, send authentication in the initial REGISTER rather than waiting to send it after receiving a 401 Unauthorized message. Defaults to 0. |
Session Timer | Boolean, Defaults to 0 | If enabled, the phone will send SIP session timers throughout the call, ending the call when there is no reply. Defaults to 0. |
S Timer Expires | Integer, in seconds | Sets the SIP Session timer timeout value in seconds. |
Enable GRUU | Boolean, Defaults to 0 | If enabled, the phone will append GRUU information to the Contact header of INVITEs. Defaults to 0. |
DTMF Mode | Integer, 0-3, Defaults to 3 | Sets the DTMF method to be used by the phone, as follows: Inband - 0 |
DTMF Inf oMode | Integer, 0-1, Defaults to 0 | If set to 1, and the phone is configured for SIP INFO DTMF, the * and # keypresses will send "*" and "#" respectively. If set to 0, and the phone is configured for SIP INFO DTMF, the * and # keypresses will send as 10 and 11 respectively. Defaults to 0. |
NAT Type | Boolean, Defaults to 0 | If enabled, STUN will be used. Defaults to 0. |
Enable Rport | Boolean, Defaults to 0 | If enabled, the phone will send rport to assist with NAT traversal as per RFC3581. Defaults to 0. |
Subscribe | Boolean, Defaults to 0 | If enabled, the phone will subscribe for Message Waiting Indicator (MWI). Defaults to 0. |
Sub Expire | Integer, in seconds, Defaults to 3600 | The timer at half-of-which, the phone will re-subscribe. Defaults to 3600. |
Single Codec | Boolean, Defaults to 0 |