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VoIP Phones (PY-90,PY-60,PY-65,PL-18)
-How to use
"Auto Provisioning" function?
-How many servers may PY-90 register simultaneously?
-Why the settings vanish after reboot?
-How to use the dial rule?
-How to use speed dial function?
-How to use Call Forward, Call Transfer and 3-way?
-Can I print my logo on the voip phones ?
Asterisk card (Model : X400P)
-What is the difference between loopstart, groundstart, and kewlstart signalling?
-How to build and customize an IP PBX with Asterisk ?
-Does it support Asterisk?Need a special driver ?
Asterisk card (Model : RX-1E1)
-How to select E1 or T1?
-Does it support 3.5 V or 5V?
-Does it support Asterisk?Need special driver ?
SIP ATA (Model : GN-06)
-How to get the ip address of the sip ata GN-06?
-How to config the dial plan of the sip ata GN-06?
VoIP Phones (PA1688)
-How
to use "Call Transfer" function?
-How
to use "Call Waiting" function?
-How
to use "Conference" function?
-Why
does the IP address show 0.0.0.0?
-What
is Dial Plan used for?How to configure it?
-What
is "inner line"?
-What
is "digit map"? How to configure it?
-dual
mode"and "dual mode prefix".
-Why does it take so long before the phone logins to the server?
How to speed it up?
-How
to recover the phone if it crashed during upgrade?
-How
to set TOS field?
Q: How to use auto provisioning?
Top
Re:Click
here to download the PDF file. 
Q: How many servers may PY-90 register simultaneously?Top
PY-90 is able to register two SIP servers simultaneously, and redundancy servers. User can configure the dial peer to route calls between these servers. Please refer ¡°How to use the dial rule?¡± for detail.
Q: Why the settings vanish after reboot?Top
Please go to Config Manage?Save Config to save your setting always.
Q: How to use the dial rule?Top
PY-90 provide flexible dial rule, with different dial-rule configure, user can easily implement the following function:
----Replace, delete or add prefix of the dial number.
----Make direct IP to IP call
----Place the call to different servers according the prefix.
You can click ¡°Add¡± to add a new dial rule. Below is the detail setting of the dial-rule:
Phone Number: The Number suit for this dial rule, can be set as full match or prefix match. Full match means that if the number user dialed is completely the same as this number, the call will use this dial-rule. Prefix match means that if prefix of the number that the user dials is the same as the prefix, the call will use this dial-rule, to distinguish from the full match case, you need to add ¡°T¡± after the prefix number in the phone number setting.
Call Mode: support SIP..
Destination (optional): call destination, can be IP or domain. Default is 0.0.0.0, in this case the call will be routed to the Public SIP server. If you set the destination to 255.255.255.255, then the call will be routed to the private SIP server. Also you can key other address here to make direct IP calls
Port (optional): Configure the port of the destination, default is 5060 in SIP and 1720 for H323
Alias (optional):Set up the Alias. We support four Alias as below. Alias need to co-work with the Del
Length:
add:xxx, add prefix to the phone number, can set to reduce the dial length.
all: xxx, replace the phone number with the xxx, can use as speed dial function.
del, delete the first N numbers. N is set in the Del Length
rep:xxx£¬replace the first N numbers. N is set in the Del Length. For Example: Use wants to place a call 86755-83843088, then you can set the phone number in the dial rule as 0755T, and set the Alias as rep:86755, and set the Del Length to 3. Then all calls begin with 0755 will be changed to 86755 xxxxxxxx.
Suffix (optional):Configure suffix, show no suffix if not set
Q: How to use speed dial function?Top
There are 9 speed dial keys in the PY-90 panel, Usage:
Set speed dial number: press the speed key and enter the speed dial number and then press Menu/OK
key to save the setting.
Pick up the handset and press the speed dial key to dial the pre-define number.
Q: How to use Call Forward, Call Transfer and 3-way
Conference calls?Top
User may set up the configuration in the Call Service page to use these value add service.
Call Forward:
----Forward when busy: select Busy in the Call Forward Field, and Key in the destination phone number in the Forward Number. If some one calls you when you having a call, the caller will be forwarded to the destination number.
----Forward no answer: Select No Answer in the Call Forward Field, and Key in the destination phone number in the Forward Number, fill the time in the No Answer Time. If some one calls you and no one answer the caller during the No Answer Time, the call will be forward to the destination number.
----Forward Always: Select Always in the Call Forward Field, and Key in the destination phone number in the Forward Number, then any one call this gateway will be forward to the destination number.
Call Transfer:
Check the Enable Call Transfer.
Unattended transfer:
If A is the PY-90 user, and B calls and talking with A through VoIP. A can press FWD button to hold the call with B, and then enter C¡¯s number. B will be transferred to C and can talk with C.
Attended transfer:
If A is the PY-90 user, and B calls and talking with A through VoIP. A can press Hold button to hold the call with B, and then enter C¡¯s number to talk will C. and press Hold to switch back to A, and then press FWD key , B will be transferred to C and can talk with C.
3-Way Conference Calls
Check Enable Three Way Call
Assume A is the RG-530 user, and B calls and talking with A through VoIP. A can press FWD button to hold the call with B, then enter * and then enter C¡¯s number to talk with C, and then press * button again to make 3-way conference calls.
Q: What is the difference between loopstart, groundstart, and kewlstart signalling?Top
Re: Loopstart signalling is used by virtually all analog phone lines. It allows a phone to indicate on hook/offhook, and the switch to indicate ring/no ring.
Kewlstart is based on loopstart, but extends the protocol by allowing the switch to drop battery on the phone line to indicate to the phone that the other end of the party has disconnected the call. Most real phone switches, and almost no PBX's (except Asterisk, of course) support this feature. It is generally required for getting hangup notification.
Groundstart signalling is sometimes used by PBX's. If you don't know what it is, don't worry, you won't need it.
Q:Does it support Asterisk?Need a special driver ? Top
Re: Yes, of course. It is the same to Digium TDM400P .
It doesnot need special driver.
Asterisk card (Model : RX-1E1)
Q:How to select E1 or T1? Top
Re : There is a jumper .
Q:Does it support 3.5 V or 5V? Top
Re : Both.
Q:Does it support Asterisk?Need a special driver ? Top
Re : Yes, of course.It is the same to Digium TE110P.
It doesnot need special driver.
Q: How to transfer a call ?
Top
Re: There
are 3 ip hones, A,B and C , A is calling B, you need transfer the
call from B to C , here are the steps,
1) B pickup the call when it is ringing, 2) press the button "Transfer
" , then the LCD says "Input the number" ,3) input
the phone number of C and press "#" , Then C is ringing
and B is busy tone.
Q: How to use "Call Waiting"?
Top
Re: Press
"Hold" to keep a call, and press it again to recover.
Q: How to use "Conference"?
Top
Re: When
Phone A is talking with Phone B , Phone B press "Conference"
key, then press the number of Phone C, then A,B,C have a conference.
The VoIP server must also have the function.
Q: Why does the IP address show 0.0.0.0?
Top
Re:
Because the IP phone use the ¡°DHCP¡± but without gettint the IP address,
here are some possible reasons: Re: connection failure B: Router/IPShare/Switch
didn¡¯t start DHCP Service.
Q: What is Dial Plan used for?How to configure it? Top
Re: Dial plan is used to convert the number dialed to the number
actually sent based on "dial number", "ddd prefix", "idd prefix",
"ddd code", "idd code"these parameters. Parameter "use dial plan"contains
5 options:
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1."disable"
Number sent is the same as number dialed.
2."enable"
If the number dialed is prefixed by "idd prefix", the number
sent will be the rest part of the number dialed dropping off
the "idd prefix".
Otherwise, if the number dialed is prefixed by "ddd prefix",
the number sent will be "idd code"+ the rest part of the number
dialed dropping off the "ddd prefix".
Otherwise, the number sent will be "idd code"+ "ddd code"+ the
number dialed.
For example: set "ddd code"= 10, "idd code"=86,
"idd prefix"=00, "ddd prefix"=0.If the number dialed is 00-1-4089821818,
the number sent will be 1-4089821818. If the number dialed is
0755-83843088, the number sent will be 86755-83843088. If the
number dialed is 83843088, the number sent will be 861083843088.
3."dial number"
Convert the number dialed to a new number according to "enable"dial
plan, then prefix "dial number"to this new number. This new
number is the number sent.
4."prefix"
Effective only when "dial number"is not empty. This serves for
those calling cards.
The number sent is just "dial number". During the call connection,
program will automatically send out a number made up with ["account"+
"pin"+ number converted by "enable"dial plan +¡®#¡¯].
5."hotline"
The number sent is "dial number". Whenever the handset is lifted
up, this number will be automatically dialed out. |
Q: What is "inner line"? Top
Re: Parameters "inner line"and "inner line prefix"help to distinguish
whether the number dialed is an inner line number. "inner line"contains
3 options:
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1."disable"
No distinction among numbers. The number sent is the number
converted according to parameter "use dial plan".
2."enable"
If the number dialed is prefixed by "inner line prefix", the
number sent is just the number dialed because this number is
an inner line number. Otherwise, the number sent is the number
converted according to parameter "use dial plan".
3."omit prefix"
The only difference between 2 "enable"is that if the number
dialed is prefixed by "inner line prefix", the number sent is
the number dialed dropping off this prefix. |
Q: What is "digit map"? How to configure it?
Top
Re: Digit map is a set of rules to determine when the user released dialing. With digit map, users don't have to press ¡®#'key
or "call"key after dialing. Digit map comes from MGCP protocol. In
MGCP, Call Agent will send a digit map to endpoint, so "use digit
map"is always selected and users don't have to store a digit map file
in the endpoint beforehand. With other protocols, a digit map file
has to be stored in the phone beforehand and can be upgraded by pressing
the [update digitmap] button or click the [upgrade firmware] at the
bottom of the settings web page. The default digit map file used in
our program is stdmap.txt. For
detailed information, please refer to RFC3435 section 2.1.2. Here
is a brief description of some items in stdmap.txt:
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X
represents any number between 0 and 9.
T represents a period of time after the user pressed the last
number. Default value is 5 seconds. [0-5]: Any number between
0 and 5. That is 0, 1, 2, 3, 4, 5.
13xxxxxxxxx: Any 11 digits number starting with 13.
02[0-57-9]xxxxxxxx: Any number starting with 02, the second
number between 0 and 5, the third number is between 7 and 9,
and total length is 12.
*x.T:Starting with * key, and users have waited T seconds after
pressing the last digits.
x.T: Any digit number and users have waited
T seconds after pressing the last digits.
If the number dialed matches some item in the digit map file,
or it doesn't match with any item, this number will be sent
out immediately. Not like "use dial plan", "use digit map"won't
change the number dialed, the number sent is the same as the
number dialed. "use digit map"can be combined with
"use dial plan"and "inner line". First using digit map to determine
when the user finished dialing, then convert this number to
the number actually sent according to "use dial plan"and "inner
line". |
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Q: "dual mode"and "dual mode prefix". Top
Re:If
"dual mode"is set to "pstn first", users can call any PSTN number
directly; If users want to call an IP phone, please pat the
hook once or press the flash key and wait until the dial tone
is heard, now the phone is ready to make an IP call.
If "dual mode"is set to "IP first", users can make an IP call
directly; If users want to call a PSTN phone, please dial the
number set in "dual mode prefix"field first and wait until the
dial tone is heard, now the phone is ready to make a PSTN call.
In both cases, the phone will ring no matter the incoming call
is from PSTN network or IP network. Users can pick up the handset
to accept the call.
If "dual
mode"is set to "disable", this 1S1O gateway is reduced to a
1S port gateway.
If powered off, the gateway will be automatically switched to
the PSTN network so that users can make PSTN calls as usual.
Q: Why does it take so long before the phone logins to the server?
How to speed it up? Top
Re:Before registration, phone will first visit FTP server and
check if the server has got new firmware or settings to upgrade, then
it will visit the time server on the Internet to get current date
and time. If the FTP server's address is a domain name, program will
visit DNS to resolve this domain name into IP address. Besides, if
local IP address is allocated by DHCP, we will check if the allocated
IP address conflicts with other devices. All these work will cost
some time to complete, which causes the delay of registration. To
speed it up, try the following two methods:
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1.set
"upgrade type"to "disable", then phone will not check new upgrades
from the FTP server.
2.set "sntp ip"to "255.255.255.255"， then phone will
not visit timer server to get date and time. If one must know
the current date and time, please set "sntp ip"to the IP address
of a local time server which has fast access. |
Q: How to recover the phone if it crashed during upgrade?
Top
Re:1.Press
and hold * while power up, until you see * begin to display
on your LCD.
2.Power off and do 1 again, after the second time power up,
our standard design IP phone's default IP address is 192.168.1.100.
3.Put the ip phone and a Windows PC on the same LAN.
4.Run palmtool.exe on PC. You can try palmtool
"Start Debug" button, a debug window will pop us, and if you
press key on the IP phone, the key and the IP phone's IP address
will be displayed on the debug window
5.If you PC's address is not 192.168.1.xxx (not on the same
LAN of the IP phone), change its TCPIP settings to the same
LAN.
6.Put 192.168.1.100 (or other default IP) in the palmtool "IP
Address on Chip" field.
7.Find the correct upgrade file, and use palmtool "Update Program"
button to update your IP phone.
Q: How to set TOS field? Top
Re:The 8-bit TOS field specifies how the datagram should be
handled and is broken down into five subfields: Let's assume that
bit 0 is the most-important-bit and bit 7 is the least-important-bit.
Bit 0-2: Three PRECEDENCE bits specify datagram precedence, with values
ranging from 0 (normal precedence) through 7 (network control), allowing
senders to indicate the importance of each datagram.
Bit 3: Also called D bit. When set, the D bit requests low delay.
Bit 4: Also called T bit. When set, the T bit requests high throughput.
Bit5: Also called R bit. When set, the R bit requests high reliability.
Bit 6-7: Unused
TOS is just a hint to the routing algorithm that helps it choose among
various paths to a destination. An internet does not guarantee the
type of transport requested.
Q: How to get the ip address of GN-06?
Top
Re:Press the keys "**00" ,then press "20#" , you can hear the ip address .
Q: How to config the dial plan of GN-06? Top
Re:.The Dial Plan is a set of rules to determine when the user released dialing, it tells your sip ATA what numbers are allowed to be dialed. It also allows you to do things like 7-digit dialing. In the Dial Plan field below, we've entered a generic Dial Plan that will work with the majority of providers. It allows you to dial an arbitrary number of digits, dial * then any number of digits, dial ** then any number of digits, and # then any number of digits). If you want a North American-style dialplan that offers 7-digit dialing, 11-digit dialing for North America calls, and standard PSTN-style international dialing (dial 011, country code, area code and number), cut and paste the following dialing plan into the Dial Plan field. You will need to then adjust the plan as follows:
Replace 1415 with your Country Code (US = 1) and Area Code (415). For example, to use US area code 360, replace 1415 to 1360.
Replace 011 with your international access code.
(<:1415>[2-9]xxxxxx|011xx.|1[2-9]xx[2-9]xxxxxx)
If you are in the US and want 10-digit dialing instead of 7-digit dialing:
(<:1>[2-9]xx[2-9]xxxxxx|011xx.|1[2-9]xx[2-9]xxxxxx)
The dial plan used for Broadeon Communications is:
(*[*x]x|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)
The dial plan used for swissipcom is:
(0[1-9]xxxxxxxx|00xxxxxxxxxxx.|[2-9])
The BroadVoice sanctioned dial plan is:
(*xx|#xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)
The dial plan used for Annatel is:
(0[1-9]xxxxxxxxS0|00[1-9]x.|26[1-9]x.|*xx.|x|*xx*x.|*xxx*x.)
The dial plan used for ATP is:
(*xx | <:TOMB>000 | <:OTMB>13xxxx | <:02>xxxxxxxx | xxxxxxxxxx | xxxxxx | xxxxxxxxxxxxxxxxxx | 190*.!)
The dial plan used for iTele is:
(xxxxxx|112|00x.|*6*xxxxxx|*6*00x.|*xx)
The iConnectHere sanctioned dial plan is:
(*xx*xxxxxxxxxxx.|*x*xxxxxxxxxxx.|[3469]11|0|00|[2-9]xxxxxx|xxxxxxxxxxxx.)
A suggested dial plan for Musimi is:
(112|00[0-9].|*xx|*31*00x.|*31*xxxxxxxx|xxxxxxxxS0)
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Service 