Grandstream wave ошибка 404

I am getting an error like :sox: Unknown output file format for ‘ring1.ring’:File type ‘ring’ is not know.”

– Rename sox.solaris to sox
– Execute the command like this: ./sox xxx.mp3 ring1.ring

What is an Outbound proxy? Should I put an Outbound proxy in the field?

An Outbound proxy is mostly used in presence of a firewall/NAT to handle the signaling and media traffic across the firewall. Generally, if you have an outbound proxy and you are not using STUN or other firewall/NAT traversal mechanisms, you can use it. However, if you are using STUN or other firewall/NAT traversal tools, do not use an outbound proxy at the same time.

Can I configure SKYPE or IAX protocol on any Grandstream device?

No, unfortunately both SKYPE and IAX use a proprietary protocol while all Grandstream products follow the SIP protocol Version 2.0 RFC 3261.

Do Grandstream products support IAX/IAX2?

IAX/IAX2 are proprietary protocols and Grandstream does not currently support these protocols.

For BudgeTone 100 Series

BT101 and BT102 models

Off-hook the receiver or press speakerphone.
Press the MENU button
Now dial the IP Address in 12 digit format ex.192168001029
Press SEND.

Note: You will need to have SIP Server field blank, along with NAT traversal set to NO, no STUN Server configured and Use Random Ports set to NO. You can make calls between public IPs and Private IPs under the same LAN.

For GXP2000/GXV3000/BT200 Models

From firmware version 1.1.0.13 onwards, these models have the ability to dial an IP address of another endpoint under the same LAN segment by simply pressing the last octet in the IP address.

In the Advanced Settings page there is an option “Use Quick IP-call mode”, by default it is set to No. When this option is set to YES, and #XXX is dialed, where X is 0-9 and XXX <=255, phone will make direct IP call to aaa.bbb.ccc.XXX where aaa.bbb.ccc comes from the local IP address REGARDLESS of subnet mask.

#XX or #X are also valid so leading 0 is not required (but OK).

eg.
192.168.0.2 calling 192.168.0.3 just dial #3 follow by SEND or #
192.168.0.2 calling 192.168.0.23 just dial #23 follow by SEND or #
192.168.0.2 calling 192.168.0.123 just dial #123 follow by SEND or #

192.168.0.2 dial #3 and #03 and #003 has same effect –> call 192.168.0.3

Note: If you have a SIP Server configured, Direct IP-IP call will still work. However, if you are using STUN, Direct IP-IP call will also use STUN.

Public IP-IP Calling for GXP2000/GXV3000

– Press the round MENU button to enter the GUI.
– Scroll down to Direct IP Call and press the round button to select this option.
– Move the cursor to the blank space above the OK and CANCEL using the round button.
– Enter the IP Address ex. 192*168*1*29
– Press the round button once you finish entering, to move the cursor to OK.
– Press the round button final time to initiate the IP-IP call.
– Public IP-IP Calling for BT200

Off-hook the receiver or press speakerphone.
Press the MENU button
Now dial the IP Address in 12 digit format ex.192168001029
Press SEND.

For GXP21xx/14xx/11xx Models

– For GXP21xx/14xx/116x, press MENU button to bring up main menu ang select “Direct IP Call” to enter the Direct IP Call mode. For GXP110x, press *** to enter IVR menu. Then enter 47 for Direct IP Calling. A dial tone will be heard again.

– Input the 12-digit target IP address. For example:If the target IP address is 192.168.1.60 and the port is 5062 (i.e., 192.168.1.60:5062), input the following: 192*168*1*60#5062.The * key represents the dot (.), the # key represents colon (:). Wait for about 4 seconds and the phone will initiate the call.

– Press the “More” softkey to make sure the softkey selection “IPv4” or “IPv6” is correctly selected depending on your network environment.
– Press “OK” softkey to dial.
– When “Use Quick IP Call Mode” is set to “Yes” under Web GUI->Advanced Setting page, users could make quick IP Call on the phone. Steps below:

– Take the phone off hook.
– Dial #xxx where x is 0-9 and xxx<255.
-Press # or SEND and a direct IP call to aaa.bbb.ccc.XXX will be completed. “aaa.bbb.ccc” is from the local IP address regardless of subnet mask. The number #xx or #x are also valid. The leading 0 is not required (but it’s OK).

– For example:
192.168.0.2 calling 192.168.0.3 — dial #3 followed by # or “SEND”;
192.168.0.2 calling 192.168.0.23 — dial #23 followed by # “SEND”;
192.168.0.2 calling 192.168.0.123 — dial #123 followed by # “SEND”;
192.168.0.2 dialing #3 and #03 and #003 results in the same call — call 192.168.0.3.

Note: 1. The # will represent colon “:” in direct IP call rather than SEND key as in normal phone call.2. If you have a SIP server configured, direct IP call still works. If you are using STUN, direct IP call will also use STUN.3. Configure the “User Random Port” to “No” when completing direct IP calls.

For GXP2200 Model

– Off hook the phone or select account in the idle screen to bring up call screen on GXP2200.
– Under “Mode Select” button in the call screen, tap on it and select the mode as “IP Call”.
– Input the 12-digit target IP address. For example, if the target IP address is 192.168.1.60 and the port is 5062 (i.e., 192.168.1.60:5062), input the following: 192*168*1*60#5062 (The * key represents the dot (.) and the # key represents the colon (:).)
– Press “SEND” key or tap on “SEND” button to dial.

Note:
1. The # will represent colon “:” in direct IP call rather than SEND key as in normal phone call.
2. If you have a SIP server configured, direct IP call still works. If you are using STUN, direct IP call will also use STUN.
3. Configure the “User Random Port” to “No” when completing direct IP calls.

For HandyTone Series

Pick up the receiver and press to access the Voice Prompt.
On hearing the Prompt press 47 for Direct IP-IP call.
Now, simply press the IP Address in 12 digit format ex. 192168001029
Press # or the Send button.

Note: You will need to have SIP Server field blank, along with NAT traversal set to NO, no STUN Server configured and Use Random Ports set to NO. You can make calls between public IPs and Private IPs under the same LAN.

How can I upload a Ringtone on my Grandstream phone?

Linux Version:
Please use the Linux link provided as our phones will only use the format generated by our tools. How to use the tool is attached in the readme file. Be advised that only .wav files can be converted. If you have .mp3 files, they need to be converted to the .wav file format. You can find a free conversion tool online. The ringtone file can not be more than 100KB each. We suggest less than 70KB each. BT100 phone will only take two ring tone files and GXP2000 can take 3 ring tone files.

Windows Version:
Please use the Windows link provided HERE
There are 8 steps to follow using the ringtone generator:

• Upload the ringtones to a computer.
• Save the file as a .wav file.
• Select the appropriate ringtone generator.
• Open the .exe file and then select (and double click) the ringtone file from your files.
• Click “generate ringtone file”. A .bin file will be created in the folder where the .wav file is located.
• Put the ring1/ring2/ring3.bin file under HTTP/TFTP server.
• Enter HTTP/TFTP server IP address in “Firmware Server Path” on Advanced Settings page of web UI.
• Click on UPDATE and REBOOT the device, just like how we do firmware upgrade.
• Then, the phone connects to the HTTP/TFTP server and grabs the ring tone file. After the phone downloads the file, the ringtone can be changed by using keypad or web UI. The config tool doesn’t support .mp3 files.

How do I point my device to a HTTP/TFTP Server?

There are 2 ways to set up your HTTP/TFTP server. Use the device web configuration page. Login to the web configuration pages of your unit. Under Advanced Settings page you will find Firmware Server Path field. Enter the IP Address or server path here. Use the phone keypad. For BudgeTone Phones: From phone keypad, press menu button and down arrow to item number 6, press menu button one more time to get into the “Editing” mode. If the TFTP server IP displayed is not the one you want, enter the entire 12 digits of the IP address of your TFTP server. e.g., if your TFTP server is 192.168.1.100, enter 192168001100. For GXP2000/GXV3000: From phone keypad, press MENU button (round button), scroll down to Config–>Upgrade to enter the Firmware Server information. For ATAs: Unhook the receiver, press ***, press 13, enter IP Address of the Firmware Upgrade Server in 12 digit format ex. 168075215189. Use option 15 to select upgrade protocol (TFTP or HTTP).

How do I setup my Grandstream Phone for Delta3/iconnect network?

Typical configuration is:

SIP Server: natrelay.deltathree.com outbound proxy: leave it blank User ID: xxxxxx (your Delta3 account number) Authentication/Login ID: xxxxx (same as above, your Delta3 account number) Password: xxxxx (your Delta3 password) Dial plan: 6666

How do I setup my Grandstream Phone for FWD service?

Typical configuration is:

SIP Server: fwd.pulver.com
Outbound proxy: 192.246.69.247:5082 (used only when behind firewall, otherwise leave it blank)
User ID: xxxxxx (your FWD account number)
Authentication/Login ID: xxxxx (same as above, your FWD account number)
Password: xxxxx (your FWD password)
NAT Traversal: No (You need to set up your STUN server if you don’t have outbound proxy)

How do I setup my Grandstream Phone for go2call network?

Typical configuration is:

SIP Server: voip01.go2call.com
Outbound proxy: (Should leave it blank, because it’s a GW)
User ID: xxxxx (your Go2Call PIN number)
Authentication ID: same as your User ID
Password: xxxxxxx (Your Go2Call password)
NAT Traversal: YES (WITHOUT setting the STUN server)

How do I setup my Grandstream Phone for MCI(test) network?

Typical configuration is:

SIP Server: siptest.mci.com
Outbound proxy: (use an outbound proxy if MCI provides one for you)
User ID: xxxxx (your MCI assigned account/phone number)
Authentication ID: (Your MCI assigned id, i.e., foo)

Password: your MCI password
NAT Traversal: No (You need to set up your STUN server if you don’t have outbound proxy)
Note: MCI Proxy server seems to respond our phone client SIP messages correctly.

How do I setup my Grandstream Phone for nikotel network?

Typical configuration is:

SIP Server: calamar0.nikotel.com
Outbound proxy: leave it blank
User ID: xxxxx (your nikotel account number)
Authentication ID: same as your User ID
Password: your nikotel password
NAT Traversal: YES (WITHOUT setting the STUN server)

How do I setup my Grandstream Phone for SIPphone.com service?

Typical configuration is:

SIP Server: proxy01.sipphone.com
Outbound proxy: (leave it blank)
User ID: xxxxxx (your SIPphone.com account number)
Authentication/Login ID: xxxxx (same as above, your SIPphone.com account number)
Password: xxxxx (your SIPphone.com password)
NAT Traversal: Yes (You need to set up your STUN server if you don’t have outbound proxy)
STUN Server: stun01.sipphone.com

How do I setup my Grandstream Phone for Telappliant VoIPtalk service?

Typical configuration is:

SIP Server: voiptalk.org
Outbound proxy: (use an outbound proxy if VoIPtalk provides one for you)
User ID: xxxxx (your VoIPtalk assigned account/phone number)
Authentication ID: (Your VoIPtalk assigned id)
Password: your VoIPtalk password

NAT Traversal: No (You need to set up your STUN server if you don’t have outbound proxy)

How do I setup my Grandstream Phone for VoIPBuster service?

You need a VoIPBuster username and password in order to call via VoIPBuster. Simply sign up and create your login at www.VoIPBuster.com. If you need help configuring your SIP device, please check the manual that came with the Grandstream SIP phone

Typical Configuration is:

SIP Server: sip1.voipbuster.com
User ID: Your VoIPBuster username
Auth ID: same as User ID
Auth Password: Your password
STUN Server: stun.voipbuster.com
Domain/Realm (optional): voipbuster.com

Codecs: G711 (64 kbps), G726 (32 kbps), G729 (8 kbps), G723 (5.3 & 6.3 kbps).
If you have audio problems:

Use a STUN server (if supported by your device) Use the G.711 codec

How do I setup my Grandstream Phone for Yesfone service?

Typical configuration is:

SIP Server: sip.yesfone.net
Outbound proxy: leave it blank
User ID: xxxxx (your Yesfone account number)
Authentication ID: same as your User ID
Password: your Yesfone password
NAT Traversal: YES (WITHOUT setting the STUN server)

How do you setup daylight savings for all products?

US Time Zones:

Pacific: PST+8PDT+7,M3.2.0,M11.1.0
Mountain: MST+7MDT+6,M3.2.0,M11.1.0
Central: CST+6CDT+5,M3.2.0,M11.1.0
Eastern: EST+5EDT+4,M3.2.0,M11.1.0
Note: For GXP2200 that doesn’t have self-defined time zone option, please select the matching time zone under time zone selection list.
Pacific Example:
Time Zone
Since US is west of the Prime Meridian, it will use “+” positive for the offset. Pacific Standard Time (PST) offset is UTC -8:00, so you will change that to PST+8. If you look up PST day light savings, you will find out that it will change to Pacific Daylight Time (PDT) and will move one hour forward (Fall/Autumn BACK, Spring FORWARD). The PDT offset is UTC -7:00, so you will use PDT+7. Daylight Savings
Before we continue, make sure to add a “,” comma. The rest of the syntax is adding the dates where daylight savings takes place. First section is for when daylight savings starts. Second section is for daylight savings ends. “M” is for month and “3” is March, 3rd month of the year. The next number is for the number of the week which is “2”, the second week of the Month. Last number is “0” which stands for the first day of the week which is Sunday. Sunday =0,Monday=1, Tuesday =2, etc. Add another comma “,” to separate between starting and ending daylight savings. Daylight savings ending starts with “M11” which is November, followed by “1” (first week) and “0” (Sunday). If you look up daylight savings US, you’ll find out that it starts on March 10th, (3rd month, 2nd week on Sunday = M3.2.0) and ends on November 3rd (11th month, 1st week on Sunday = M11.1.0).

Is there additional equipment required for my headset to work?

Refer to HERE for the compatible headset list of GXP16xx series.

Refer to HERE for the compatible headset list of GXP2130/ GXP2135/ GXP2140/ GXP2160/ GXP2170 and GXP17xx.

Refer to HERE for the compatible headset list of GXV3240 & GXV3275.

Based on the headsets that we’ve tested we found the below combinations of headsets and quick disconnect cables that work great together on our phones.

Based on the headsets that we’ve tested we found the below combinations of headsets and quick disconnect cables that work great together on our phones.

*Note: All corded headsets require a specific quick disconnect cable.  Please see below to find out what cord is needed with your Plantronics, Jabra, Ovislink, or VXI headset.

What codec should I use for my Grandstream phone?

PCMU (G711u) is used by default. Both PCMU and PCMA will give you toll quality but their bandwidth consumption is also the highest. If your network bandwidth is low, you can choose a lower-bit-rate codec such as G723 or G729 which will give you near toll quality at much smaller bandwidth consumption. If bandwidth is not a concern try using PCMU or PCMA, or even the wideband codec G722 which will provide hi-fidelity voice.

For example: If you have a basic high-speed connection at home (768kbps/128kbps), configure your phone with either G723 or G729 to ensure best available voice quality. If you have T1 bandwidth capacity, configure your phone with either G711 (a/µ) or G722 to experience toll-quality or high-fidelity voice.

What data range we should put for Layer 2 / Layer 3 QoS?

Layer 2 QoS – 802.1q VLAN 802.1p priority
VLAN tag has 12 bits so the value range is from 0 – 4095. 0 means no VLAN
Priority value has 3 bits so the value range is from 0 – 7

Layer 3 QoS – DSCP
It has 6 bits with the range from 0 to 63.

*Note: All corded headsets require a specific quick disconnect cable.  Please see below to find out what cord is needed with your Plantronics, Jabra, Ovislink, or VXI headset.

What if my SIP URI domain is different from the SIP proxy server FQDN (Fully Qualified Domain Name)?

With firmware 1.0.3.60 and later, you can put the your SIP URI domain name into the SIP Server field, and put the actual sip server FQDN into Outbound Proxy field. The phone will use the domain name in SIP Server as part of SIP URI but send and receive SIP messages through the SIP proxy server defined in the Outbound Proxy field.

What is Early Dial? Should I use it?

When you dial a number, if you do not press the “Dial” ( “Redial”) or “#” key if it is configured to function as the “Send” key at the end of your dialed string, the phone will wait for about 4 seconds before timeout and then sends the actual INVITE message. If you set “Early Dial” to be YES, then the phone will attempt to send out INVITE at each key input using the entered dial string collected so far. If the SIP server supports 484 Incomplete Address response, the phone will keep trying with each new key entry until the complete dialed string is entered. This will essentially eliminate the 4-second wait time mentioned above.

Please note that this option can be used ONLY when the SIP server supports 484 Incomplete Address response. Otherwise, any other negative responses from the SIP server (such as 404 Not Found) will cause immediate termination of the call.

What is Early Dial? Should I use it?

STUN stands for Simple Traversal of UDP over NAT. It is a protocol which enables an IP phone to detect the presence and type of NAT behind which the phone is placed. An IP phone that supports STUN can intelligently modify the private IP address and port in its SIP/SDP message by using the NAT mapped public IP address and port through a series of STUN queries against a STUN server located on the public Internet. This will allow SIP signaling and RTP media to successfully traverse a NAT without requiring any configuration changes on the NAT.

STUN presents a working solution for most NATs that are not symmetric NAT, e.g., most of the SOHO routers have non-symmetric NAT and in this case, it is OK to use STUN. However, STUN does NOT work with symmetric NAT and if your routers have built-in symmetric NAT, do not use STUN.

Note: NOT ALL SIP PROXY SERVERS WILL WORK WITH A STUN TRANSLATED SIP MESSAGES, PLEASE CONSULT YOUR SERVICE PROVIDER FOR DETAIL.

What is the difference between “User ID” and “Authentication ID”?

User ID is the user part of the SIP address of the phone and this is usually the information displayed as Caller ID on the LCD. e.g., typically it is a phone number or extension number or a user’s name. Authentication ID is an ID used strictly for authentication purpose when the phone attempts to contact the SIP server. This may or may not be the same as User ID.

What is the RMA procedure?

All RMA’s are handled by the Grandstream Certified Partner that sold the product(s). Additional information can be found : here

What is TR-069 and do Grandstream devices support it?

It is a protocol for communication between CPE (Customer Premise Equipment) and an ACS (Auto Configuration Server) that provides secure auto-configuration as well as other CPE management functions within a common framework. Grandstream devices do support this protocol.Click here for more details.

What number should I use for “Voice Frames per TX”?

It depends on what codec you choose and balance between bandwidth utilization and impact of packet loss. The bigger this value, thehigher bandwidth utilization because more voice frames are packed into the payload field of a UDP/RTP packet and thus the network header overhead would be lower. However, the impact of a packet loss on perceived voice quality will be bigger.

• For PCMU/PCMA, the default is 2 and max is 10
• For G723, the default is 1 and max is 32
• For G726-32, the default is 2 and max is 20
• For G729, the default is 2 and max is 64
• For G728, the default is 4 and max is 64

What precautions must I take before performing an upgrade?

Firmware upgrade is a simple yet crucial procedure. If done incorrectly, it could damage the unit. On our website we have list of instructions and warnings that should be read carefully before attempting an upgrade. You can find this information here (enter your product reference to get all necessary documents).

What type of headset connections are available on my Grandstream phone?

What voice codecs do Grandstream products support?

All the phones support similar codecs and are listed below. The BT100 does not support GSM and the GXV does not support G722. Each codec has its uniqueness for certain application.

HT Series – PCMU (G711u), PCMA (G711a), G729A/B, G723.1, G726-32, iLBC
HT502 – PCMU (G711u), PCMA (G711a), G729A/B/E, G723.1, G726-40/24/16, iLBC
BT Series – PCMU (G711u), PCMA (G711a), G729A/B, G723.1, G722, G726-32, iLBC
BT200 supports PCMU (G711u), PCMA (G711a), G729A/B, G723.1 and GSM codecs.
GXP Series – PCMU (G711u), PCMA (G711a), G729A/B, G723.1, G722, iLBC
GXV3000 – PCMU (G711u), PCMA (G711a), G729A/B, G723.1, GSM
GXP2200 – PCMU (G711u), PCMA (G722a), G729A/B, G722

Where can I download the sample Ringtone Pack?

Users can download the sample ringtone pack from the link HERE. Follow the instructions above (Steps 6-8) to upload the sample ringtone to the phone.

Where can I find the Firmware Version of my Grandstream device?

There are two ways that you can see your unit’s firmware version depending on the product:

For ALL Grandstream devices
The software version is displayed on top of the Status web configuration page in a format similar to the following:

Software Version: Program–1.0.X.X Bootloader–1.0.X.X HTML–1.0.X.X The number series next to Program indicates the firmware version of the unit.

For BudgeTone Series Phones
Press “menu” button
Press up arrow button 3 times to get to item number 9 “codE rEL”
Press “menu” one more time to browse the “codE rEL”
Use up or down arrow button to see “B (bootloader)” and “P (program)” code date and version.
For GXP21xx/GXP14xx/GXP116x/GXP2000/GXV3000
Press MENU button
Scroll down to Status option and select it by pressing the MENU button again
Scroll down to Boot and Prog. Prog is the firmware version on the device.

Where can I find user manuals for all products?

Click here and select your Grandstream product. You will now have all resource for this product including manuals

Which EHS headset works on my GXP2200, GXP2160 and GXP2140 phone?

For full support of EHS, you will need the following cable and headset models from Plantronics:

Headsets:

CS500 Series (CO52,CO54)

Savi 700 Series (W101/W100,W740)

EHS Adapters:

APD-80 (If this is the EHS adapter that was purchased, you will also need the 85638-01 or 40287 cable in order to work.)

Plantronics also has a communication hub(headset base only) model MDA200 that is also support on our phones.

Note: The EHS adapter is not included with the headsets nor the communication hub. GXP2200, GXP2160, GXP2140, GXP2124 and GXP116x are the only models that support EHS headsets.

Refer to HERE for the compatible headset list of GXP2130/ GXP2135/ GXP2140/ GXP2160 & GXP2170.

Refer to HERE for the compatible headset list of GXV3240 & GXV3275.

Which Grandstream phone supports Bluetooth headsets and which headset brand/model does it support?

The GXP2200, and the list of bluetooth headsets that have been tested and verified as working:
Plantronics – M25, M55, M100, M155, M165, M1100, BackBeat Go, Calisto 620
Motorola – H681, H720, HX550, HX730, HK200
Samsung: HM1000, HM1200, HM1700, WEP490
Jabra: EasyGo, PRO 9470
Sony Ericsson – MW600, HPM-78
LG – HBM235
Logitech – UE3100
Emerson – Over the Head EM237C
Nokia: BH-214, BH-108
Powerblue – 712
VXI – Blueparrott B250XT+
VXI – Blueparrott Xpressway II<

Refer to HERE for the compatible headset list of GXP16xx series.

Refer to HERE for the compatible headset list of GXP2130, GXP2135, GXP2140, GXP2160 and GXP2170.

Refer to HERE for the compatible headset list of GXV3240 & GXV3275.

Which NTP server can I use?

By default, the NTP server is set to either “time.nist.gov” or “us.pool.ntp.org”. If that or your own NTP server does not work, try to select an NTP server from following link: http://www.eecis.udel.edu/~mills/ntp/servers.html or you can find more info from www.ntp.org.

Support & resources