SIP Extension VOIP

Last Updated : Jun 20, 2024 |
Navigation: Call ManagementExtensionsEdit ExtensionSIP VoIP
Prolog information
Resource Id: extension103
These settings are shown for SIP IP extensions. For example for J100 Series phones.
Field
Description
IP Address
Default = 0.0.0.0
The IP address of the phone. If an address is entered, the IP Office only accept registration from a device with that address.
Reserve License
Default = None.
Avaya IP phones require an Avaya IP Endpoint license, non-Avaya IP phones require an 3rd Party IP Endpoint license. Normally the IP Office issues licenses in the order that devices register. This option allows this extension to be pre-licensed before the device registers. This can prevent a previously licensed phone becoming unlicensed following a system restart. The options are:
  • Reserve Avaya IP Endpoint License
  • Reserve 3rd Party IP Endpoint License
  • Both
  • None
Note:
  • When WebLM licensing is enabled, this field is automatically set to Reserve Avaya IP Endpoint License. The Both and None options are not available.
  • When the Profile of the corresponding user is set to Centralized User, this field is automatically set to Centralized Endpoint License and cannot be changed.
VoIP Silence Suppression
Default = Off
When enabled, when the IP Office detects silence during an IP call, it does not send any audio data.
  • Lines between IP Office systems using G.711 ignore this feature.
  • On trunks between networked IP Office systems, you must enable the setting at both ends.
Fax Transport:
Default = Off.
This option is only available if Re-Invite Supported is selected. When enabled, the system performs fax tone detection on calls routed via the line and, if fax tone is detected, renegotiates the call codec as configured below. The SIP line provider must support the selected fax method and Re-Invite.
For systems in a network, fax relay is supported for fax calls between the systems.
The options are:
  • None - Select this option if fax is not supported by the line provider.
  • G.711 - Use G.711 to send and receive faxes.
  • T38 - Use T38 to send and receive faxes.
  • T38 Fallback - Use T38 to send and receive faxes. If the call destination does not support T38, the IP Office will send a re-invite to change the transport method to G.711.
DTMF Transport
Default = RFC2833.
This setting is used to select the method by which DTMF key presses are signalled to the remote end. The supported options are In Band, RFC2833 or Info.
Requires DTMF
Default = Off.
This field is displayed when System SettingsSystemVoIPIgnore DTMF Mismatch for Phones is set to On. It can be used to allow direct media connections between devices despite the devices having differing DTMF setting.
When Requires DTMF is set to Off, during the checks for direct media, the system ignores the DTMF checks if the call is between two VoIP phones. Set to On if the extension needs to receive DTMF signals.
SIP endpoints using simultaneous login, which do not have physical extensions in the configuration, are treated by the system as not requiring DTMF.
Note:
  • Direct media may still not be possible if other settings, such as codecs, NAT settings, or security settings, are mismatched.
Local Hold Music
Default = Off.
When enabled, the extension plays local music when on HOLD.
If System SettingsLineAdd/Edit Trunk LineSIP LineSIP AdvancedLocal Hold Music is enabled, the extension Local Hold Music must be disabled to play far end music to the extension.
Allow Direct Media Path
Default = On
This settings controls whether calls between IP endpoints and/or lines must go through the IP Office.
  • If disabled, calls go through the IP Office and use its resources. RTP relay allows calls between devices using the same audio codec to not require a voice compression channel.
  • If enabled, calls can take routes other than through the IP Office system. Both ends must support direct media and have matching VoIP settings. For example, both ends must use the same protocol (SIP or H.323), same addressing (IPv4 or IPv6), and so on. Otherwise, the call goes through the IP Office system.
    • For extensions, disabling Requires DTMF allows the extension to attempt direct media even if the other end has differing DTMF settings.
VoIP Silence Suppression
Default = Off
When enabled, when the IP Office detects silence during an IP call, it does not send any audio data.
  • Lines between IP Office systems using G.711 ignore this feature.
  • On trunks between networked IP Office systems, you must enable the setting at both ends.
Codec Lockdown
Default = Off.
After a SIP offer with a list of codecs, the IP Office expects an SDP answer with a single codec selected from that list. However, some SIP user agents send an SDP answer with multiple codecs. Those user agents then expect to switch to any of the codecs during the call without further negotiation. IP Office does not support this and loss of speech path will occurs if the user agent changes codec without renegotiating.
  • If enabled, when the IP Office receives an SDP answer that contains multiple codecs from the list of codecs it offered, the IP Office sends a re-INVITE and a SIP offer with just a single codec.
  • This option requires Re-Invite Supported enabled.
3rd Party Auto Answer
Default = None.
This setting applies to 3rd party standard SIP extensions. The options are:
  • RFC 5373: Add an RFC 5373 auto answer header to the INVITE.
  • answer-after: Add answer-after header.
  • device auto answers: IP Office relies on the phone to auto answer calls.
Media Security
Default = Same as System.
These settings control whether SRTP is used for this extension and the settings used for the SRTP. The options are:
  • Same as System: Matches the system setting at System SettingsSystemVoIP Security.
  • Disabled: Media security is not required. All media sessions (audio, video, and data) is enforced to use RTP only.
  • Preferred: Media security is preferred. Attempt to use secure media first and if unsuccessful, fall back to non-secure media.
  • Enforced: Media security is required. All media sessions (audio, video, and data) is enforced to use SRTP only. Selecting Enforced on a line or extension that does not support media security results in media setup failures
    • Calls using Dial Emergency switch to using RTP if enforced SRTP setup fails.
Codec Selection
Default = System Default
Set the supported codecs. Where possible, Avaya recommend that you use the same set of codecs for all IP Office systems, lines, and extensions.
The options are:
  • System Default - Use the codec list set in the system settings.
  • Custom - Configure a list of codec preferences for the line.
    • You can move codecs between the Unused and Selected sets and change the order of the codecs.
    • The codecs available are set by System SettingsSystemVoIP. The possible codecs are:
      • OPUS - Supported on Linux-based IP Office systems only.
      • G.711 ALAW/G.711 ULAW
      • G.729
      • G.723.1 - Supported on IP500 V2 systems only.
      • G.722 64K - Supported by Linux-based IP Office systems and on IP500 V2 systems with IP500 VCM, IP500 VCM V2 or IP500 Combo cards.