Aastra Telecom 800 User Manual
Voice over IP (VoIP)
SIP Telephony
68
information is available for NAT on an incoming RTP connection. To avoid this
problem, the IP address of a workstation computer or telephone visible on the
Internet is determined using a STUN server (STUN: Simple Traversal of UDP over
NAT). You can ask your SIP provider for the STUN server’s IP address and port
number. If you don’t need a STUN server, leave the SIP Provider field empty.
problem, the IP address of a workstation computer or telephone visible on the
Internet is determined using a STUN server (STUN: Simple Traversal of UDP over
NAT). You can ask your SIP provider for the STUN server’s IP address and port
number. If you don’t need a STUN server, leave the SIP Provider field empty.
■
For direct SIP telephony using Aastra 800, only SIP IDs consisting of numbers
for identifying subscribers registered with the SIP provider specified can be
addressed
for identifying subscribers registered with the SIP provider specified can be
addressed
■
You can integrate an external SIP connection in the Telephony: Trunks: Route
menu into the route configuration. You can use a network provider rule to
specify the routing of numbers within a specific range to use SIP telephony as a
preference (see also PBX Networking, under Configuration starting on page 94).
menu into the route configuration. You can use a network provider rule to
specify the routing of numbers within a specific range to use SIP telephony as a
preference (see also PBX Networking, under Configuration starting on page 94).
You can configure SIP connections in the Configurator on the pages Telephony:
Trunks: SIP trunks and Telephony: Trunks: SIP provider. Enter the technical
attributes of a specific SIP provider, such as the IP addresses for the registrar and
the STUN server under SIP provider. Under SIP trunks enter the information for
an existing SIP account, such as the user name, password, assigned call number
and the maximum number of simultaneous calls possible.
Trunks: SIP trunks and Telephony: Trunks: SIP provider. Enter the technical
attributes of a specific SIP provider, such as the IP addresses for the registrar and
the STUN server under SIP provider. Under SIP trunks enter the information for
an existing SIP account, such as the user name, password, assigned call number
and the maximum number of simultaneous calls possible.
7.3.2 Internal SIP Subscribers
The Aastra 800 becomes available as the SIP server for internal SIP subscriber
telephony switching services. SIP telephones connected via LAN or SIP programs
installed on workstation computers can thus establish connections to all other
subscribers or trunks connected to the Aastra 800.
telephony switching services. SIP telephones connected via LAN or SIP programs
installed on workstation computers can thus establish connections to all other
subscribers or trunks connected to the Aastra 800.
License Assignment
The number of possible SIP subscribers is determined by the number of licenses
purchased. In order to provide you with the greatest possible flexibility regarding
usage of available licenses, license assignment is dynamic via the “Floating
license”. Using a user/password combination (“SIP log on”) you can have several
SIP subscribers under the same call number. Only every new SIP log-on occupies a
new license. The technical log-on process of a SIP subscriber with a valid user
name and correct password is always successful. Only when a connection is estab-
lished is there an attempt made to occupy a license under the SIP log-on. If all
licenses are occupied currently, the SIP subscriber can only make emergency calls.
The number of possible SIP subscribers is determined by the number of licenses
purchased. In order to provide you with the greatest possible flexibility regarding
usage of available licenses, license assignment is dynamic via the “Floating
license”. Using a user/password combination (“SIP log on”) you can have several
SIP subscribers under the same call number. Only every new SIP log-on occupies a
new license. The technical log-on process of a SIP subscriber with a valid user
name and correct password is always successful. Only when a connection is estab-
lished is there an attempt made to occupy a license under the SIP log-on. If all
licenses are occupied currently, the SIP subscriber can only make emergency calls.