Yes. In addition a setting of 1-8 rings before auto-answer can be selected.

 Symetrix VoIP Interface – Network Diagnostics

CISCO or other third party device manufacturers have no control of the Symetrix VoIP cards. Therefore the settings will have no bearing on Symetrix VoIP cards performance.

Yes, all lines connected to the system via multiple Symetrix VoIP cards can be combined. Each line must first be registered, but the combining of the lines is done within Composer software with mixing and routing.

Dialing can be done from Composer software, as well as many other methods. Third party devices such as AMX or Crestron can be used for dialing via Ethernet (UDP and TCP-IP) or RS-232.  SymVue software can also be used for dialing.

VLANS can be used but only one VLAN can be used per card, which applies to both lines because there is only 1 IP per card.

The same settings apply to both lines per card. 802.1Q User Priority can be set from 0-7, as well as DSCP settings for Audio RTP and Call Control.

The Codec settings are assigned for each card, so the same settings apply to both lines on a card.

The Symetrix VoIP card has 2 lines, with 2 call appearances per line. Each line gets its own extension. Only 1 IP address is assigned per card.

Yes, the Symetrix VoIP card works the same as any other VoIP phone on the network.

The MAC address can be found in Composer software and on the Web admin.

The Symetrix VoIP card must be used with a SIP based proxy server.

No, only IPv4 is supported at this time.

Yes, the VoIP card is set to DHCP by default. It can easily be assigned to a static IP if needed from Composer software.

First the IT department needs to establish the extension number. Then it is entered in the VoIP card’s web-based administration interface.

No, the Symetrix VoIP card doesn’t require a PoE switch as it is powered from the DSP unit it is installed into. If a PoE switch is used, it will not damage the Symetrix VoIP card.

Limited only by the hosting DSP hardware type and numbers used. A single Edge Frame can host up to 4 cards, while Radius AEC and Radius 12xEX can each host a single card.

This is user configurable in the VoIP card’s web-based administration interface using a simple drag and drop method.

G.722, G.711 (A-law and μ-Law), G.729 (including Annex B) and G.723.1. Note that G.729 Annex B is currently supported for received audio only, the full implementation of VAD for all codecs will be in a future release.

The Symetrix VoIP card has 2 lines per card. 4 call appearances per VoIP card (4 remote calls, 2 call appearances per line).

Our products utilize SIP 2.0 to interconnect with VoIP servers. We found that while SIP is a standard, there are many variances in the way it is applied by VoIP providers. SIP 2.0 is a loose specification that allows the developers of SIP-based systems to make many decisions on features and functions. The consequence of this is that two SIP based systems can be completely SIP 2.0 compliant and completely incompatible.

We have gone to great lengths to optimize our solutions for market-leading providers. Symetrix’s VoIP Interface Card has been tested for interoperability with Cisco, Avaya, 3CX and Asterisk and we intend to continue to test interoperability with other systems.

Much of the setup and configuration of a VoIP system is out of Symetrix’s and the AV integrator’s control. This includes the configuration and components of the VoIP network. We will make every effort to support the configuration of our VoIP Interface Card but if there are issues with the VoIP Interface Card not functioning with systems for which we have not created a configuration tech note, we can provide guidance but troubleshooting will need to be done with the assistance of the enterprise IT department or telephony system provider and interoperability cannot be guaranteed. We are unable to assist with the configuration of VoIP PBX systems.