Does the GXW series support RFC 3262?
Yes. GXW400x supports RFC 3263 – DNS resolution.
Does the GXW series support RFC 3264?
Yes. GXW400x supports RFC 3264.
Does the GXW series support RFC 3515?
Yes. GXW400x supports RFC 3515 – REFER method.
Does the GXW series support RFC 3960?
Yes. GXW400x supports RFC 3960 – early-media / ringback.
How can I configure different settings for different channels?
Check the syntax for your required setting. For example, if you want all incoming PSTN calls to be automatically dialed to the same extension 200 for an auto-attendant call without a handset, use the syntax: “ch1-8:200;”. However, if you prefer to route channel 1 to extension 200, channels 2-7 to extension 201, and channel 8 to extension 202 for an open auto dial, use the following syntax: “ch1:200;ch2-7:201;ch8:202;”.
How can I troubleshoot fax issues on my GXW?
To troubleshoot fax issues, we recommend the following:
Avoid using low-bandwidth codecs like G729. This codec component almost does not detect fax tones (latest f/w has improvement in this area).
You can force a codec to be a certain codec by setting an initiate code on the fax machine (*027110 for PCMU, for example).
Always use T38. Ensure your PBX supports and enables this protocol.
Some fax machines don’t auto negotiate speeds. It is recommended to force the baud rate to 9600 (even if HT500 supports 14.4 bps).
How many simultaneous calls can GXW4024 handle?
GXW4024 will support 24 concurrent calls for all codecs.
How many simultaneous fax calls can GXW40xx handle?
The GXW40xx series can handle up to two concurrent T.38 sessions and G.711 pass-through on the remaining ports.
If multiple DIDs from the service provider are packaged with only one trunk main account, how should I configure the gateway to route calls to different FXS ports?
Grandstream GXW IP gateways support call routing based on SIP Request URI User ID. This function can help route calls to different FXS ports within a hunt group.
For incoming calls:
Enter SIP trunk ID, password, and Name information to an FXS port, and save it. Then enable Hunt Group under FXS Ports -> Advanced Port Settings.
Assign other FXS ports to the enabled hunt group. Then, name them with the extra DIDs packed in your sip truck, in the Request URI Routing ID field.
For outgoing calls: After completing step 2 above, go to Profile X -> SIP Settings -> Basic Settings, and set Use Request Routing ID in SIP Headers to “Yes”.
Once done, the gateway will look at the User ID in the INVITE of the incoming call, and route the call to the matching FXS port associated with the corresponding DID. Also, the gateway will replace the From and To headers in the INVITE with the configured Request URI Routing ID for outbound calls.
What is a dial plan?
A dial plan is a set of rules that govern a device’s call routing behavior. When a user dials a number sequence, the device refers to the rules in the dial plan to determine how to route the call best. For more details on how to configure a dial plan, please refer to the GXW410x User Manual in our resource library.
What is dial pause in the dial plan, and how can I configure it on my GXW410x?
In some offices, employees may need to dial ‘9’ before dialing a phone number. If a user dials ‘9’ + phone number as a single consistent string, the call won’t go through, and a general prompt like “Your call cannot be completed as dialed. Please check the number and dial again.” will be generated. This can be easily resolved by adding a dial pause in the dial plan.
To add a dial pause to the dial plan, you will need to configure the “DTMF Inter Digit (X10ms) Dial Pause” under the “Dialing Plan” tab in the web GUI. The syntax used to configure the pause is: ‘ch1-4:d2p200, d4p400; section5-8: d1p100, d3p300’ where dx/py means a 10y-ms pause after the d-th digit is dialed.
If you need a 100ms pause after the 4th digit is dialed, the syntax will be ch1-4:d4p10.
If you need to add a 300ms pause after the 3rd digit is dialed and a second 400ms pause after the 6th digit, the syntax will be: ch1-4:d3p30,d6p40.
Note: Dial plan pause works ONLY in Stage 1 dialing method.
What is the difference between one-stage dialing and two-stage dialing?
One-stage dialing means the end user hears the dial tone immediately and can start dialing.
Two-stage dialing means the end user needs to dial twice – once to reach the second dial tone and once more to reach the final destination. In other words, a call transitioning from PSTN to VoIP or VoIP to PSTN needs to go through two dialing stages to reach the intended recipient. For instance, a VoIP user will complete the first stage by dialing a pre-programmed number (on GXW410x), receiving the PSTN dial tone in return, and then dialing a PSTN number to complete the second stage. The reverse happens for PSTN-to-VoIP calls.
With one-stage dialing, if a call is made by VoIP or PSTN, it goes through the gateway to the intended recipient. With two-stage dialing, a call transitioning between PSTN and VoIP needs to go through two dialing stages to reach its final destination. For PSTN-originated VoIP calls, it goes directly to a VoIP extension.
One-stage dialing provides convenience and seamless calling, while two-stage dialing offers flexibility and full access to the VoIP network.
I can receive incoming PSTN calls, but I cannot make outgoing calls (one-stage or two-stage). Why?
You might be experiencing compatibility issues with some PSTN lines such as Verizon, Qwest, or others.
The following issues might occur:
One-stage VoIP to PSTN calls result in immediate flashing of the line LED.
Two-stage VoIP to PSTN calls are not picked up by GXW410x.
There is no caller ID on incoming PSTN calls (if caller ID is available).
To resolve the above issues:
Contact Grandstream Support or visit www.grandstream.com for the latest firmware.
Configure the following values on the FXO Lines web configuration page:
Enable Disconnect Current After Dial to “Yes” (if PSTN provider is using Disconnect Current).
Current Disconnect Threshold: 300
Pre-Dial Min Delay PSTN: 750
Why doesn’t my GXW410x receive incoming PSTN calls using one-stage dialing?
Ensure that you have a valid internal extension set in the VoIP Unconditional Call Forwarding setting on the FXO Lines web configuration page. Also, make sure that this extension is accessible by the GXW410x. For example, if you’re using Asterisk™ and have a SIP account with identity 200, but there is no extension 200 in the context where the GXW410x resides, the GXW410x won’t be able to reach this SIP user, and incoming calls from PSTN won’t proceed.