Basic Settings

DescriptionYou can enter a name for the SIP provider.
Provider StatusSelect whether this VoIP provider entry is enabled
Active:
Inactive:
Access Configuration

Select which type of VoIP phone numbers you wish to configure.

 Single Number(s)Enter the individual DSL phone numbers.
Direct Dial-In :Enter a basic number in conjunction with an extension number block.
Authentication IDEnter your provider's authentication ID.
PasswordAt this point, you can assign a password.
Username 
Domain 

Outgoing Signalisation Settings

Outgoing Signalisation

Select the signal you want for outgoing calls.
Global CLIP no Screening NumberOnly for Outgoing Signalisation Global CLIP no Screening Number
Signal remote caller numberOnly for Outgoing Signalisation = Global CLIP no Screening Number and Individual CLIP no Screening Number
Signal fixed out numberOnly for Outgoing Signalisation = Fixed Out DDI

Registar

RegistrarEnter the DNS name or IP address of the SIP server. A 26 digit alpha-numeric sequence is possible.

Registrar Port

Enter the number of the port to be used for the connection to the server. The default value is 5060 . A 5 digit sequence is possible.
Transport ProtocolSelect the transport protocol for the connection.
UDP 
TCP 

STUN

STUN serverEnter the name or the IP address of the STUN server.
Port STUN serverEnter the number of the port to be used for the connection to the STUN server.

Timer

Registration TimerEnter the time in seconds within which the SIP client must re-register to prevent the connection from disconnecting automatically.

 

 

Advanced Settings

ProxyEnter the DNS name or IP address of the SIP server. A 26 digit alpha-numeric sequence is possible.
Proxy PortEnter the number of the port to be used for the connection to the proxy. The default value is 5060 . A 5 digit sequence is possible.
Transport ProtocolSelect the transport protocol for the connection.
UDP  
TCP  

Further Settings

From Domain

Enter the SIP provider's "From Domain". It is used after the @ as sender data in the SIP header of the SIP data packages.

 

The value entered in From Domain will replace the value entered in Domain.

Note: this only applies to the From field.

Request-Line: INVITE sip:0628548111@provider.com;transport=udp SIP/2.0
Message Header
Via: SIP/2.0/UDP 192.168.2.13:5060;branch=z9hG4bK9A8B83B951C4311082C5378012300025;rport

 From: <sip:0314760249@FromDomainExample;user=phone>;tag=84BC6AE7CAC53110AC9E378012300025
SIP from address: sip:0314760249@FromDomainExample
SIP from address User Part: 0314760249
SIP from address Host Part: FromDomainExample
SIP tag: 84BC6AE7CAC53110AC9E378012300025
To: <sip:0628548111@provider.com;user=phone>
SIP to address: sip:0628548111@provider.com
Call-ID: 48B558DCCBC53110ACBB378012300025
CSeq: 2 INVITE
Contact: <sip:hybirdtest@192.168.2.14:5060;transport=udp;line=84BC6AE7CAC53110AC9E378012300025>
Contact Binding: <sip:hybirdtest@192.168.2.14:5060;transport=udp;line=84BC6AE7CAC53110AC9E378012300025>
Max-Forwards: 70
[truncated] Proxy-Authorization: Digest username="hybirdtest", uri="sip:0628548111@provider.com;transport=udp", realm="sipserver", nonce="537f07d3000044e8c670b983efc6c79f2baab7a31fcd6a1c", algorithm=MD5, response="45e4501a3c20d647
Authentication Scheme: Digest
Username: "hybirdtest"
Authentication URI: "sip:0628548111@provider.com;transport=udp"
Realm: "sipserver"
Nonce Value: "537f07d3000044e8c670b983efc6c79f2baab7a31fcd6a1c"
Algorithm: MD5
Digest Authentication Response: "45e4501a3c20d6475bf37f704634935b"
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, MESSAGE, SUBSCRIBE, UPDATE, PRACK, REFER
Supported: 100rel, replaces, timer
User-Agent: Hybird_120_GE V.9.1 Rev. 8 (Beta 5) IPSec
Allow-Events: refer, message-summary, dialog
Session-Expires: 1800
Content-Type: application/sdp
Content-Length: 333

Number of allowed simultaneous CallsSelect the maximum number of calls that shall be simultaneously possible Please also note the settings for bandwidth management here.
Location

Select the location of the SIP server. Locations are defined in the VoIP->Settings->Locations menu.
Any Location(default value)The server is not operated at any defined location.
<Location Name>Defined in the VoIP->Settings->Locations menu.
Codec ProfilesSelect the codec profile for this SIP server. Codec profiles are defined in the VoIP->Settings->Codec Profiles menu.
 System Default(default value) The server is operated with a codec profile predefined in the system.
 <Codec profile name>Defined in the VoIP->Settings->Codec Profiles menu.
Dial End Monitoring TimeSelect the time (after dialling the last digit of a call number) after which the system begins external dialling.
Call Hold inside the PBX systemSelect whether a telephone call in the system can be switched to hold without losing the connection (inquiry calls/brokering). If this function is not enabled, the call is held at the SIP provider, if he supports this performance feature.
Call Forwarding extern (SIP 302)Select whether calls are to be redirected externally with the SIP provider. The call is forwarded using SIP status code 302.
Generate international phone numberIf you have enabled this function and entered 43 (for Austria) underGlobal Settings Country Profile, the 0043 before a number dialled with area code is generated automatically.
Generate national subscriber numberIf you have enabled this function and entered the National Prefix / City Code ( e.g. for Abfaltersbach 4846 ) under Global Settings, the 4846 before a number dialled with area code is generated automatically.
Deactivate number suppressionIf you enable this function, the number is always sent, independently of whether you have switched Suppress outgoing CLIP (CLIR) on or off for an extension.
SIP Header Field for User NameSelect the position of the user name (user ID) in the SIP header for outgoing calls.
None

The username is not transmitted.

P-Asserted

The so-called "p-asserted-identity" field is added to the SIP header and contains the User Name.

The values that will be used in the P-asserted identity fields are the values entered in:
VOIP->Settings->Basic configuration Username and Domain.
However if an address has been entered in From domain this will be used instead of the value entered in the Domain field.

Request-Line: INVITE sip:0628548111@provider.com;transport=udp SIP/2.0
Message Header
Via: SIP/2.0/UDP 192.168.2.13:5060;branch=z9hG4bK9A8B83B951C4311082C5378012300025;rport
From: <sip:hybirdtest@provider.com>;tag=84BC6AE7CAC53110AC9E378012300025
To: <sip:0628548111@provider.com;user=phone>
Call-ID: 6639FCFBCAC53110ACA4378012300025
CSeq: 2 INVITE
Contact: <sip:hybirdtest@192.168.2.14:5060;transport=udp;line=84BC6AE7CAC53110AC9E378012300025>
Max-Forwards: 70
[truncated] Proxy-Authorization: Digest username="hybirdtest", uri="sip:0628548111@provider.com;transport=udp", realm="sipserver", nonce="537f065a000185b6875115b4dd608a8ba94ce631b8f623d1", algorithm=MD5, response="157b626f9e241753
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, MESSAGE, SUBSCRIBE, UPDATE, PRACK, REFER
Supported: 100rel, replaces, timer
User-Agent: Hybird_120_GE V.9.1 Rev. 8 (Beta 5) IPSec
Allow-Events: refer, message-summary, dialog
P-Asserted-Identity: <sip:hybirdtest@provider.com;user=phone>
Session-Expires: 1800
Content-Type: application/sdp
Content-Length: 333

P-Preferred

The so-called "p-preferred-identity" field

is added to the SIP header and contains the User Name.

The values that will be used in the P-asserted identity fields are the values entered in:

VOIP->Settings->Basic configuration Username and Domain.

However if an address has been entered in From domain this will be used instead of the value entered in the Domain field.

Request-Line: INVITE sip:0628548111@provider.com;transport=udp SIP/2.0
Message Header
Via: SIP/2.0/UDP 192.168.2.14:5060;branch=z9hG4bKA49A7273CAC53110AC91378012300025;rport
From: <sip:hybirdtest@provider.com>;tag=F2BBB822CAC53110AC86378012300025
To: <sip:0628548111@provider.com;user=phone>
Call-ID: D2776773CAC53110AC8F378012300025
CSeq: 2 INVITE
Contact: <sip:hybirdtest@192.168.2.14:5060;transport=udp;line=F2BBB822CAC53110AC86378012300025>
Max-Forwards: 70
[truncated] Proxy-Authorization: Digest username="hybirdtest", uri="sip:0628548111@provider.com;transport=udp", realm="sipserver", nonce="537f057500015a0f24dde5226593e4cd99b485b59015dd4b", algorithm=MD5, response="b971394dfbdc9908
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, MESSAGE, SUBSCRIBE, UPDATE, PRACK, REFER
Supported: 100rel, replaces, timer
User-Agent: Hybird_120_GE V.9.1 Rev. 8 (Beta 5) IPSec
Allow-Events: refer, message-summary, dialog
P-Preferred-Identity: <sip:hybirdtest@provider.com;user=phone>

SIP PPI Address: sip:hybirdtest@provider.com

SIP PPI User Part: hybirdtest

SIP PPI Host Part: provider.com
Session-Expires: 1800
Content-Type: application/sdp
Content-Length: 333


SIP Header Field(s) for Caller AddressSelect the position of the sender ID (e.g. subscriber number) in the SIP header for outgoing calls. (For incoming calls, the subscriber number is taken automatically from the SIP header.)

Display

The sender ID is placed in the "Display" field of the SIP header.

The value of the Display field is determined by the value configured as Outgoing Signalisation for this user.

 This value can be configured in Numbering->User settings

Request-Line: INVITE sip:0628548111@provider.com;transport=udp SIP/2.0
Message Header
Via: SIP/2.0/UDP 192.168.2.13:5060;branch=z9hG4bK9A8B83B951C4311082C5378012300025;rport

 From: "0314760249" <sip:hybirdtest@provider.com>;tag=48DE3DF947C4311081D4378012300025
SIP Display info: "0314760249"
To: <sip:0628548111@provider.com;user=phone>
Call-ID: 427DCC3C50C43110829D378012300025
CSeq: 2 INVITE
Contact: <sip:hybirdtest@192.168.2.13:5060;transport=udp;line=48DE3DF947C4311081D4378012300025>
Max-Forwards: 70
[truncated] Proxy-Authorization: Digest username="hybirdtest", uri="sip:0628548111@provider.com;transport=udp", realm="sipserver", nonce="537c8af8000049aae2b43c36efbf7f771ea1456384929b5c", algorithm=MD5, response="76260831131f87a3
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, MESSAGE, SUBSCRIBE, UPDATE, PRACK, REFER
Supported: 100rel, replaces, timer
User-Agent: Hybird_120_GE V.9.1 Rev. 8 (Beta 5) IPSec
Allow-Events: refer, message-summary, dialog
Session-Expires: 1800
Content-Type: application/sdp
Content-Length: 333

Username

The sender ID is transferred to the "User" field of the SIP header.

The value of the sender ID is determined by the value configured as Outgoing Signalisation for this user.

 This value can be configured in Numbering->User settings

Request-Line: INVITE sip:0628548111@provider.com;transport=udp SIP/2.0
Message Header
Via: SIP/2.0/UDP 192.168.2.13:5060;branch=z9hG4bK9A8B83B951C4311082C5378012300025;rport
From: <sip:0314760249@provider.com;user=phone>;tag=48DE3DF947C4311081D4378012300025
SIP from address: sip:0314760249@provider.com
SIP from address User Part: 0314760249
SIP from address Host Part: provider.com
SIP tag: 48DE3DF947C4311081D4378012300025
To: <sip:0628548111@provider.com;user=phone>
SIP to address: sip:0628548111@provider.com
Call-ID: D2CD7CB951C4311082C3378012300025
CSeq: 2 INVITE
Contact: <sip:hybirdtest@192.168.2.13:5060;transport=udp;line=48DE3DF947C4311081D4378012300025>
Contact Binding: <sip:hybirdtest@192.168.2.13:5060;transport=udp;line=48DE3DF947C4311081D4378012300025>
Max-Forwards: 70
[truncated] Proxy-Authorization: Digest username="hybirdtest", uri="sip:0628548111@provider.com;transport=udp", realm="sipserver", nonce="537c8d200000b6bd4ff2bc632913cba1ae9cfff5ba681e00", algorithm=MD5, response="afb159fa93bca015
Authentication Scheme: Digest
Username: "hybirdtest"
Authentication URI: "sip:0628548111@provider.com;transport=udp"
Realm: "sipserver"
Nonce Value: "537c8d200000b6bd4ff2bc632913cba1ae9cfff5ba681e00"
Algorithm: MD5
Digest Authentication Response: "afb159fa93bca0156223adf55a98b63d"
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, MESSAGE, SUBSCRIBE, UPDATE, PRACK, REFER
Supported: 100rel, replaces, timer
User-Agent: Hybird_120_GE V.9.1 Rev. 8 (Beta 5) IPSec
Allow-Events: refer, message-summary, dialog
Session-Expires: 1800
Content-Type: application/sdp
Content-Length: 333

P-Preferred

The so-called "p-preferred-identity" field is added to the SIP header to transmit the sender ID there.

The values that will be used in the P-asserted identity fields are the values entered in:

VOIP->Settings->Basic configuration Username and domain

Request: INVITE sip:0628548911@provider.com;transport=udp, with session description

Frame 5 (1466 bytes on wire, 1466 bytes captured)
Ethernet II, Src: Elmegt_61:a1:ec (00:09:4f:61:a1:ec), Dst: Draytek_eb:89:40 (00:50:7f:eb:89:40)
Internet Protocol, Src: 192.168.2.14 (192.168.2.14), Dst: 89.184.172.54 (89.184.172.54)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
Request-Line: INVITE sip:0628548911@provider.com;transport=udp SIP/2.0
Message Header
Via: SIP/2.0/UDP 192.168.2.14:5060;branch=z9hG4bKA49A7273CAC53110AC91378012300025;rport
From: <sip:hybirdtest@provider.com>;tag=F2BBB822CAC53110AC86378012300025
To: <sip:0628548911@provider.com;user=phone>
Call-ID: D2776773CAC53110AC8F378012300025
CSeq: 2 INVITE
Contact: <sip:hybirdtest@192.168.2.14:5060;transport=udp;line=F2BBB822CAC53110AC86378012300025>
Max-Forwards: 70
[truncated] Proxy-Authorization: Digest username="hybirdtest", uri="sip:0628548911@provider.com;transport=udp", realm="sipserver", nonce="537f057500015a0f24dde5226593e4cd99b485b59015dd4b", algorithm=MD5, response="b971394dfbdc9908
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, MESSAGE, SUBSCRIBE, UPDATE, PRACK, REFER
Supported: 100rel, replaces, timer
User-Agent: Hybird_120_GE V.9.1 Rev. 8 (Beta 5) IPSec
Allow-Events: refer, message-summary, dialog
P-Preferred-Identity: <sip:hybirdtest@provider.com;user=phone>
Session-Expires: 1800
Content-Type: application/sdp
Content-Length: 333

P-Asserted

The so-called "p-asserted-identity" field is added to the SIP header to transmit the sender ID there.

The value of the sender ID is determined by the value configured as Outgoing Signalisation for this user.

 This value can be configured in Numbering->User settings

 

Request: INVITE sip:0628548911@provider.com;transport=udp, with session description

Frame 3 (1219 bytes on wire, 1219 bytes captured)
Ethernet II, Src: Elmegt_61:a1:ec (00:09:4f:61:a1:ec), Dst: Draytek_eb:89:40 (00:50:7f:eb:89:40)
Internet Protocol, Src: 192.168.2.14 (192.168.2.14), Dst: 89.184.172.54 (89.184.172.54)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
Request-Line: INVITE sip:0628548911@provider.com;transport=udp SIP/2.0
Message Header
Via: SIP/2.0/UDP 192.168.2.14:5060;branch=z9hG4bK70C3A948BFC53110AB9A378012300025;rport
From: <sip:hybirdtest@provider.com>;tag=C4A3025224C531109E73378012300025
SIP from address: sip:hybirdtest@provider.com
SIP tag: C4A3025224C531109E73378012300025
To: <sip:0628548911@provider.com;user=phone>
SIP to address: sip:0628548911@provider.com
Call-ID: 1C9FA948BFC53110AB99378012300025
CSeq: 1 INVITE
Contact: <sip:hybirdtest@192.168.2.14:5060;transport=udp;line=C4A3025224C531109E73378012300025>
Contact Binding: <sip:hybirdtest@192.168.2.14:5060;transport=udp;line=C4A3025224C531109E73378012300025>
Max-Forwards: 70
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, MESSAGE, SUBSCRIBE, UPDATE, PRACK, REFER
Supported: 100rel, replaces, timer
User-Agent: Hybird_120_GE V.9.1 Rev. 8 (Beta 5) IPSec
Allow-Events: refer, message-summary, dialog
P-Asserted-Identity: <sip:0314760249@provider.com;user=phone>
Session-Expires: 1800
Content-Type: application/sdp
Content-Length: 333

Substitution of International Prefix with "+"Select whether the prefix (e.g. 00) should be replaced by + for international numbers.
PBX couplingSelect whether another PABX can log into your system. In this way, several PABX systems can be linked.
Delete SIP bindings after RestartIf after registering with a provider a reset of the system should occur, for example, or a power failure, depending on the provider, another registration may prove impossible. Enabling these performance features allows re-registration after restart.
Upstreaming Device with NATIf you enable this function, you can use a gateway with NAT and still make VoIP calls. Without this function, it may not be possible to call you with VoIP if you use a gateway with NAT.
Early media supportSelect whether you'll allow exchange of voice and audio data before a receiver accepts a call.
Provider without RegistrationSelect whether registration and authentication with a provider can be eliminated. In this case, the relevant data can be sent to a specific IP address already known to the correspondent. An example of this method is Microsoft Exchange SIP.
T.38 FAX supportSelect whether faxes shall be transmitted with T.38.
Substitution of Incoming Number PrefixFor incoming calls, if the call number should be forwarded in the system in modified form: in the first input field enter the sequence of the incoming number to be replaced by the number sequence entered in the second input field.

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