Frequently Asked Questions

What is the default IP address of Analog Gateway?

The default IP address is 172.16.99.1

What’s the default user name and password of Analog Gateways?

Default username: admin

Password: admin

Where can I find user manuals for all Analog Gateway products?

Please check this page.

How to check Analog gateway’s IP address?

Pick up the analog phone, dialing ‘**’ can check the IP address.

Can all the FXS ports on the gateway be registered to the same VoIP Server?

Yes, you can configure one VoIP Server template and apply it to all the FXS ports.

How to connect the Analog Gateway to 3CX?

Please check this configuration guide.

How to connect the Analog Gateway to Asterisk?

Please check this configuration guide.

How to connect the Analog Gateway to Openvox UC?

Please check this configuration guide.

How to connect the Analog Gateway to Freepbx?

Please check this configuration guide.

How to connect the Analog Gateway to Isaabel?

Please check this configuration guide.

Does Analog Gateway support line reverse polarity?

Yes, you can enable polarity for the FXO port.

A polarity signal is required on some systems for reverse polarity billing.

Once receiving the signal, the system starts to bill.

What voice compression does Analog Gateway support?

Analog Gateway supports the following codecs: G.711A，G.711U，G.729，G.722，ILBC.

Which type of VPN is supported on Analog Gateway？

Openvox Analog Gateway support OpenVPN, PPTP VPN, Zerotier VPN and N2N VPN.

Does Analog Gateway support the SNMP?

Yes, Openvox Analog Gateway support SNMP. Please check the SNMP guide.

Does the Analog Gateway have any tool to debug?

Yes, our gateway has system logs and a packet tool on the Web to help you debug the gateway in real time.
Moreover, you can debug the device via SSH. You need to enable SSH on “Service” Page first.

Can I adjust the call volume?

Yes. You can adjust the call volume for the FXS/FXO port.

Go to the Channel setting page and find the volume settings.

The option“RxGain” for receive volume, and “TxGain” for transmit volume.

Increasing the RxGain value means that the internal party hears a bigger voice.

Increasing the TxGain value means that the external party hears a bigger voice.

How to configure network disconnection escape on Analog Gateway?

Please check the function guide of network disconnection escape.

What is the difference between an FXO and an FXS Gateway?

FXO Gateway – Its allows IP networks to talk to PSTN networks through the gateway by simply connecting analog lines to it. Primarily used to allow remote IP endpoints to be able to use local main office PSTN lines.

FXS Gateway – It allows your traditional telephone system to function as an IP System by simply connecting analog FXS trunks or analog handsets to the gateway. Primarily used to allow traditional handsets/PABX to be used in an IP environment.

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.

What is STUN? 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 TR-069 and do Analog Gateway 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. Analog Gateway do support this protocol.

I can make Outgoing calls but I cannot receive any incoming calls?

This is most likely due to intermediate device like Router/Firewall blocking UDP ports for SIP. Make sure you’re router or firewall has ports opened for SIP, RTP, etc like 5060, 5004. One way to check is by configuring a STUN Server (you can find free public STUN Server settings online) and then noticing the NAT type under STATUS page. If it says ‘NAT type is full cone’ you should be fine, but if it says symmetrical or port-restricted, you will need to make adjustments on the intermediate device.

Docking to a higher version of Asterisk fails

If the error shows the following message, add the from-user parameter to fix it.

NOTICE[128228]: res_pjsip/pjsip_distributor.c:673 log_failed_request: Request ‘INVITE’ from ‘”153XXXXXXX” <sip:153XXXXXXX@172.16.10.245>’ failed for ‘172.16.143.200:5060’ (callid: [email protected]) – No matching endpoint found

NOTICE[128228]: res_pjsip/pjsip_distributor.c:673 log_failed_request: Request ‘INVITE’ from ‘”153XXXXXXX” <sip:[email protected]>’ failed for ‘172.16.143.200:5060’ (callid: [email protected]) – Failed to authenticate

Add the parameter identify_by=auth_username,username to the endpoint in the server’s sip trunk.