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Grandstream Networks HandyTone HT286 Product Related Questions page 24

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Two-stage dialing means the end user has to dial twice - once to reach a second dial tone, and
again to reach the final destination. In other words, a call traversing from PSTN to VoIP or VoIP
to PSTN must go through two dialing stages to reach the intended recipient. For example, a
VoIP user will complete the first stage by calling a pre-programmed number (on the GXW410x)
and receive the PSTN dial tone in return. The user will then complete the second stage by
dialing a PSTN number. The reverse happens for PSTN to VoIP calls.
If one-stage dialing is used, a call from either the VoIP or PSTN side will pass through the
gateway to the intended recipient. With one-stage dialing, a PSTN number dialed by a VoIP user
will reach the GXW410x, and the GXW410x will immediately dial the call on the PSTN side.
For PSTN to VoIP, an incoming PSTN caller will be directly connected with a VoIP extension.
One-stage dialing offers convenient and streamlined dialing, while two-stage dialing offers
flexibility and complete access to the VoIP network.
Q: Why does the GXW410x not pick up incoming PSTN calls using one-stage dialing?
Make sure that you have a valid extension entered in the Unconditional Call Forward to
VoIPsetting under the FXO Lines web configuration page. Also, make sure that that extension is
reachable by the GXW410x. For example, if using Asterisk™ and you have a SIP account with
an ID of 200 but there is no extension 200 in the context the GXW410x resides in, the
GXW410x will not be able to reach that SIP user and will not pick up the incoming PSTN call.
Q: Why can I receive incoming PSTN calls, but outgoing calls (one-stage or two-stage)
can't grab a PSTN line?
You may have compatibility issues with some Verizon, Qwest, or certain other PSTN lines.
The following issues may occur:
One-stage VoIP to PSTN calls cause the line LED to turn on and hang up immediately
Two-stage VoIP to PSTN calls, GXW410x does not pick up a PSTN line
Incoming PSTN calls do not have caller ID information (if caller ID information is
available)
To fix the above issues:

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