Grandstream Networks HT503 user manual Way Conferencing, Pstn Pass Through

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Three situations can follow the transfer:

1.A quick confirmation tone (temporarily using the call waiting indication tone) followed by a dialtone. This indicates the transfer was successful (transferee has received a 200 OK from transfer target). A can either hang up or make another call.

2.A quick busy tone followed by a restored call (on supported platforms only). This means the transferee has received a 4xx response for the INVITE and we will try to recover the call. The busy tone indicates the transfer has failed.

3.Busy tone keeps playing. This means we have failed to receive the second NOTIFY from the transferee and the call has timed out.

Note: this does not indicate the transfer has been successful, nor does it indicate the transfer has failed. When transferee is a client that does not support the second NOTIFY (such as our own earlier firmware), this situation occurs. In bad network scenarios, this could also happen, although the transfer may have been completed successfully.

Attended Transfer

This function is applicable on the FXS port for VoIP calls only. Assume that parties A and B are in conversation. Party A wants to Attend Transfer Party B to C:

1.A presses FLASH on the analog phone to get a dial tone;

2.A then dial C’s number followed by #.

3.If C answers the call, A and C are in conversation. Then A can hang up to complete transfer.

4.If C does not answer the call, A can press “flash” back to talk to B.

NOTE: When Attended Transfer fails and A hangs up, the HT503 will ring user A back again to remind A that party B is still on the call. Party A can pick up the phone to resume a conversation with party B.

3-way Conferencing

The HT503 supports Bellcore Style 3-way conferencing.

Assume that parties A and B are in conversation. Party A (using the HT503) wants to bring C into a 3- way conference:

1.A presses FLASH (on the analog phone, or Hook Flash for old model phones) to get a dial tone.

2.A dials C’s number then # (or wait for 4 seconds).

3.If C answers the call, then A presses FLASH to bring B, C in the conference.

4.If C does not answer the call, A can press FLASH back to talk to B.

5.If A presses FLASH during the conference, C will be dropped out.

6.If A hangs up, the conference will be terminated for all three parties when configuration “Transfer on Conference Hangup” is set to “No”. If the configuration is set to “Yes”, A will transfer B to C so that B and C can continue the conversation.

PSTN Pass Through

HT503 supports PSTN pass through using the FXS port. The user can place and receive PSTN calls using analog phone connected to FXS port.

Grandstream Networks, Inc.

HT503 User Manual

Page 12 of 38

 

Firmware 1.0.4.2

Last Updated: 06/2011

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Contents Grandstream Networks, Inc Safety Compliances Warranty Table of Figures Welcome Connecting the HT503 Equipment PackagingPower LED WAN LEDLAN LED PHONE/ Line LEDSoftware Features Overview Hardware Specification LEDTo reset the HT to take affect the new IP address Understanding HT503 Voice PromptMain Menu Placing a Phone Call Phone or Extension NumbersCall Hold Call WaitingCall Transfer Way Conferencing Pstn Pass ThroughVoIP-to-PSTN Calls PSTN-to-VoIP CallsRoute Calls to Pstn Forward Calls to PstnForward Calls to VoIP One Stage DialingFax Support Enable Srtp Disable Srtp Blind TransferFlash/Hook Configuring HT503 through Voice Prompt Static IP ModeConfiguring HT503 with Web Browser Access the Web Configuration MenuNAT DNDFXS FXOMTZ+6MDT+5 DMZ IP Admin Password Firmware UpgradeLayer 3 QoS Layer 2 QoSHTTP/HTTPS ACS URLDisable Direct IP Life Line ModeConfiguration Lock KeypadAuthenticate ID Authentication PasswordDNS mode Unregister on RebootSIP T1 Timeout Disable DtmfEnable Call Features Disable Call WaitingSpecial Feature Dial Plan PrefixUse # as Dial key Dial Plan Dial Plan RulesVAD Srtp Mode Slic SettingDisable Line Echo Canceller LECAuthenticate Password Outgoing Call WithoutSIP registration failure Retry wait timeNegotiation Proxy Require Dian PlanTimeout Preferred Vocoder Invite Ring-No-AnswerCaller ID Scheme FSK Caller ID minimumRX Level dB FSK Caller ID SeizureEnable Current Enable Pstn DisconnectPstn Ring Thru Delay First Digit Timeout secSaving the Configuration Changes Rebooting from RemoteConfiguration through a Central Server Firmware Upgrade through TFTP/HTTP/HTTPS Software UpgradeConfiguration File Download Firmware and Configuration File Prefix and PostfixManaging Firmware and Configuration File Download Restore Factory Default Setting
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