After an IP phone makes an outgoing call through the AT0 trunk, the ringback tone is heard after about 8s. Why

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If the trunk function is not used, change the dialup timer value of the AT0 trunk to 0. The AR does not need to spend 5s on detecting the ringback tone. A user needs to wait for only 3s to hear the ringback tone.

Run the pbx number-parameter 40 0 command in the voice view to set parameter 40 to 0.
NOTE:
The modification may cause failures of calls transferred through trunks. The pbx number-parameter 40 0 command can be used only when POTS or SIP phones make outgoing calls through AT0 trunks.

Other related questions:
After an IP phone makes an outgoing call through the AT0 trunk, the ringback tone is heard after about 8s. Why
If the trunk function is not used, change the dialup timer value of the AT0 trunk to 0. The AR does not need to spend 5s on detecting the ringback tone. A user needs to wait for only 3s to hear the ringback tone. Run the pbx number-parameter 40 0 command in the voice view to set parameter 40 to 0. NOTE: The modification may cause failures of calls transferred through trunks. The pbx number-parameter 40 0 command can be used only when POTS or SIP phones make outgoing calls through AT0 trunks.

Why busy tone is heard after the phone is picked up
The reasons for busy tone after offhook are as follows: - The license is not correctly installed. The voice function requires the license. Apply for and install the commercial license file before using the voice function. You can run the display license state command to check whether the license is in normal state. - No PBX user is configured. This problem occurs if the device or user identifier of the PBX is not configured correctly. Therefore, ensure that device and user identifiers are correctly configured. - DSP resources are insufficient or the DSP module is not installed. Run the display voice dsp state { slot/dsp-index | channel slot/dsp-index/channel } command to check the value of the Idle parameter. (Note: In V200R002C00 and earlier versions, run the display voice dsp-dimm state { slot/dsp-index | channel slot/dsp-index/channel } command to check DSP resource usage. A later version of V200R002C00 is used as an example.) display voice dsp state 0/0 Symbol: 0-idle $-G.711 busy A-All busy W-Wastage X-fault @-IP loopback *-PCM loopback #-prohibited Channel NO. DSP channel state ---------------------------------------------------------------------- 0000-0035 00000 00000 00000 00000 00000 00000 00000 0 Total: 36 DSP channel 36 idle, 0 G.711 Busy, 0 All Busy, 0 Wastage, 0 Fault, 0 IP loopback, 0 PCM loopback, 0 prohibited The preceding information shows that there are 36 idle states, indicating that DSP resources are sufficient. If the fault persists, contact Huawei technical support personnel.

When the IP phone under the U1911 initiates an outgoing conference call through the AT0 trunk, which AT0 trunk is used to route the call?
In the scenario where the U1900 uses the AT0 trunk for networking, for a local conference initiated on the IP phone, the corresponding dedicated line is used; for an instant conference, the line is selected in polling mode. If all AT0 lines are dedicated lines, only the outgoing call for initiating a local conference can be routed through these lines.

Why the busy tone is heard after dialup
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When the U1960 uses the AT0 trunk to make outgoing calls, can the calling numbers be displayed as a fixed number?
When the U1960 uses the AT0 trunk to make outgoing calls, the calling number can be displayed as the trunk dedicated line number. Other numbers cannot be used as the calling numbers displayed to the called parties.

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