Settings > Providers and PBXs > Settings Configuration Manager Providers and PBXs

Settings Configuration Manager Providers and PBXs
Proceed as folllows:
Using the Configuration Manager (page Settings > Providers & PBXs > Configure)
Provider
Name 
Enter provider or PBX name.
 
Type 
Requirements:
Public exchange (Internet VoIP provider): Selects VoIP provider as the type.
PBX: Selects PBX as the type.
 
Domain 
Requirements:
Enter domain. The domain (also known as a "SIP domain" or "SIP realm") is required for the VoIP address. The VoIP address of a subscriber consists of the VoIP number and the domain, separated by the @ character: <subscriber>@domain.
 
Network interface type 
A network interface is an interface that enables a computer or a network component access to a computer network. Setting the network interface type allows you to operate the telephone on different networks, for example, VLANs or VPNs.
Default network: Default network is used.
VPN: VPN is used.
 
  Registrar
The registrar is the IP address or URL that you receive from your VoIP provider for login.
Important: Every port opening on the router is a security risk. Perform the appropriate security measures.
Notes:
The registrar should only be disabled if the provider requires this.
No NAT Keep Alive is performed if the registrar is disabled.
An internal network is usually protected from external manipulation by, for example, a firewall integrated in the router. Without NAT Keep Alive, the firewall's security functions will defend you against attempted accesses from the Internet.
That is why you may need to set up port forwarding for incoming SIP packets in the router for the SIP port set up in the provider's/PBX's configuration.
 
 REGISTER requests are sent to the registrar .Additional entries must be made:
Registrar address 
Enter registrar address as IP address or URL.
Registrar port (1...65535) 
Enter registar port.
Registration interval (min.) (1...60) 
Enter registration interval.
 
  Outbound proxy
The outbound proxy is an intermediate server that processes all VoIP requests and connections going to the provider (except for registration).
Requirements:
Deactivated: Disables the outbound proxy.
Received automatically: Automatically determines an outbound proxy.
Received manually: Enables you to enter an outbound proxy.
If the Received manually option has been selected, additonal entries must be made:
Outbound proxy IP/URL 
Enter outbound-Proxy IP/URL.
Outbound Proxy port (1...65535) 
Enter outbound proxy port.
Note: You find an overview of the PBX ports in the Configuration Manager of the PBX under Overviews > Ports.
 
NAT
NAT (Network address translation) converts an IP address of an e. g. private network into an IP address of the public network. All computers that communicate with each other within a private network, receive Internet access via only one IP address. The internal IP addresses of the private network are not accessible for the Internet.
  SIP NAT traversal
If NAT traversal is switched on and a query is sent from a local IP address to the public network, the sending IP address is swapped with the public IP address. This function is performed in the reverse direction for the reply.
Requirements:
STUN server set up for the provider/PBX, if the Enabled with STUN option has been selected
If the enabled option has been selected: the registrar enabled for the provider/PBX
Important: Every port opening on the NAT router is a security risk. Perform the appropriate security measures.
Enabled  : NAT traversal is performed by the provider/PBX. The SIP request also contains an "rport" part in which the IP addresses used here (local, public) are transported.
Disabled  : NAT traversal is not performed by the provider/PBX. The router, which connects the Local Area Network with the Internet, should be a properly functioning, SIP-aware router which performs NAT traversal.
Enabled with STUN  : NAT traversal is performed by the provider/PBX. To achieve this, you must also enter a STUN server under Settings.
Note: If Enabled is selected: if there are problems with unilateral call connections, a STUN server should be used (enabled when you are using STUN).
 
  RTP NAT traversal
If NAT traversal is switched on and a query is sent from a local IP address to the public network, the sending IP address is swapped with the public IP address. This function is performed in the reverse direction for the reply.
Requirements:
STUN server set up for the provider/PBX, if the Enabled with STUN option has been selected
If the enabled option has been selected: the registrar enabled for the provider/PBX
Important: Every port opening on the NAT router is a security risk. Perform the appropriate security measures.
Disabled  : NAT traversal is not performed by the provider/PBX. The router, which connects the Local Area Network with the Internet, should be a properly functioning, SIP-aware router which performs NAT traversal.
Enabled with STUN  : NAT traversal is performed by the provider/PBX. To do this, you must also specify a STUN server.
 
  NAT keep alive
Keep alive maintains the connection between client and server by sending NAT keep alive packets which prevent the interruption of the connection. An intveral for the sending of those NAT keep alive packets must be determined. This interval indicates after how many seconds NAT keep alive packets are send to maintain the NAT mapping in the firewall.
 NAT keep alive is enabled and an additional entry must be made:
 NAT keep alive interval (sec.) (15...255)
Enter NAT keep alive interval.
Note: Some providers lock accounts if the NAT keep alive interval (sec.) is too short. As a rule, this is reported with error message 503 during SIP registration. If this problem occurs, we recommend you reduce the value (e.g. to 180).
 
 
STUN server
A STUN server provides subscribers on a private network information on request. This information includes the IP address and port outside their private network that is considered as the source of their data. This information is entered in the requests, instead of the actual private IP address/port.
 STUN server IP/URL
Enter STUN server IP/URL.
 STUN server port (1...65535)
Enter STUN server port.
 STUN server query interval (min) (1...60)
Enter interval.
Requirements:
Note: If there are problems with unilateral call connections, enabling the RTP might help.
Important: Each port forwarding is a security risk. For this reason, we recommend you use this function only if absolutely necessary.
 
 
 
SIP
SIP port (1...65535) 
The SIP port is a port on the local system that is used as a starting point for the SIP transfer.
Notes:
The SIP port must be different for each provider/PBX.
You find an overview of the PBX ports in the Configuration Manager of the PBX under Overviews > Ports.
 
Enter SIP port.
 
SIP session timer  
 The SIP session timer regularly verifies if the call connection still exists. (The telephone/PBX cannot recognize if the provider interrupts the call connection).
SIP session timer interval (min.) (5...6) 
Indicates after how many minutes the SIP session timer should check the call connection.
Note: The enabling of the SIP session timer can cause increased breaks in conversations after the set interval, if a provider has not implemented the renewal of the session cleanly. In this case, you should disable the SIP session timer.
 
Server type 
Default SIP server: Server for the normal system telephone operation.
Broadsoft server: Server for providers with Broadsoft certification.
ESTOS CSTA (ECSTA): Server for the use of uaCSTA.
 
Session Initiation Protocol Security (SIPS)  
Requirements:
Important: If you are using this on the PBX, the PBX specifies the encryption method. The PBX overwrites manual settings.
 Overhearing of VoIP conversations will be prevented, as these connections are encrypted. The connection setup and termination, and also signalling, are encrypted with SIPS. The call data is encrypted with SRTP.
Note: If you have saved a root certificate for encrypting calls in your PBX, this is uploaded automatically to the telephone. You must enter the root certificate fingerprint in the telephone so that the root certificate can be verified.
Set PBX Account: Fingerprint for SIPS certificate . Enter fingerprint.
Set SIP account: Import certificate . Import certifcate from the hard disk.
Import certificate
 Import
Import certificate.
 Delete
Delete certificate.
URL structure 
SIPS: Encryption of SIPS via TLS.
SIP/TLS: Encryption of SIP via TLS.
 
Disable host name check  
 No verification whether the certificate belongs to the domain/IP.
  SIP transport protocol
UDP (User Datagram Protocol) is used to send data packet over non-secure communication lines without a connection. This means that successful transmission is dependent on the application and is therefore not always guaranteed. UDP itself does not verify whether data has been transmitted successfully. When a UDP packet is sent, the sender cannot assume that the packet will indeed arrive at the recipient. This particular protocol needs only a small amount of additional information, and results in a better data throughput rate in a well-functioning network, e.g., on a LAN. UDP is used, e.g., for the DNS (Domain Name Server).
TCP (Transmission Control Protocol) is a transport protocol that segments data into packets up to a specified size and reliably sends these individual data packets in the correct sequence to the recipient address. In this process, every data packet sent must be resent until it has been confirmed as arrived. In order to make sure this happens, a large amount of information is sent along with the actual payload data. Most Internet services are implemented with TCP, e.g., HTTP (WWW), SMTP/POPS (e-mail), etc.
Important: If encryption is enabled by SIPS, the TCP transport protocol is used. Manual settings are overwritten.
 UDP: UDP is used.
 TCP: TCP is used.
 
RTP
DTMF signalling  
At DTMF signalling, a sound of two overlapped sound frequencies is assigned to each digit of the keypad. Here, you must select how the DTMF signalling is to be transmitted. This is critically depened on the provider and the transmission path.
Note: If your are unsure which DMFT signalling type you must select, you cand find out the right type by trying them out.
Outband, with local response in acc. with RFC2833  : The telephone uses different channels to transfer DTMF signals and voice data. The DTMF signals are filtered out of the voice data. You hear a confirmation tone to confirm that the DTMF signals have been sent successfully.
Inband, DTMF signals sent through the audio channel  : The telephone uses the same channel (DTMF tones) to transfer DTMF signals and voice data.
Outband, in acc. with RFC2833  : The telephone uses different channels to transfer DTMF signals and voice data. The DTMF signals are filtered out of the voice data.
 
Codecs  
A codec is a method that encodes (digitises) analogue voice data for transmission and again decodes again, meaning converts into back into voice. There are various codecs that feature different voice data compression rates thereby require different band widths for data transmission. The equality of VoIP calls is dependent on the codec used.
Notes:
If a codec with high bandwidth (e.g. G.711) produces too much interference in the call quality, the connection bandwidth might not be sufficient. If these types of disruptions happen frequently, we recommend you select codecs with a lower band width.
Not every codec is supported by every provider.
 
Best available quality  : Codec G.722 has priority level 1, Codec G.711 has priority level 2 and Codec iLBC has priority level 3.
Best possible compression  : Codec iLBC has priority level 1, Codec G.722 has priority level 2 and Codec G.711 has priority level 3.
Force G711  : Codec G.711 is required.
 
Music on hold  
Requirements:
 When putting a caller on hold, music on hold will be played.
 
Jitter buffer (ms) (40...160) 
The size of the jitter buffer shows how many RTP packets can be buffered, to bridge or compensate disruptions.
Jitter buffer size entry in milliseconds (duration of the audio signal).
Lower values: A lower number of RTP packets can be cached in order to buffer disruptions or to compensate for them.
Greater values: A greater number of RTP packets can be cached in order to buffer disruptions or to compensate for them.
 
SRTP 
To prevent anyone from eavesdropping VoIP conversations, you can encrypt these connections. The connection setup and termination, and also signalling, are encrypted with SIPS. The call data is encrypted with SRTP.
Important: If you are using this on the PBX, the PBX specifies the encryption method. The PBX overwrites manual settings.
Disabled: Forces de-activation of call data encryption by SRTP. The connection is not established if the call partner (VoIP provider, other PBX in the sub-system operation, external VoIP subscriber) requires call data to be encrypted by SRTP.
Preferred: Switches on negotiation for the encryption of call data by SRTP. When a call is made, the call partner will be asked if encryption is possible. If so, the call data will be encrypted and then transmitted, if not, the call data will not be encrypted.
Mandatory: Forces activation of call data encryption by SRTP. The connection is not established if the call partner (VoIP provider, other PBX in the sub-system operation, external VoIP subscriber) does not support the encryption of call data by SRTP.
 
Numbers
  Use of public exchange subscriber numbers
Public exchange subscriber numbers must be used for PBXs which allow you to dial external telephone numbers without preceding exchange line access numbers or characters (e. g. 0 or **). Such PBXs automatically detect whether exchange line access is required or not.
Requirements:
 Call acceptance rules for internal and external will be applied when a call comes in. In addition, you can forward an incoming call to an external telephone number without having to dial the exchange line access number.
 
Exchange line access number 
Requirements:
Selected type PBX (page Settings > Providers & PBXs > Provider)
Note:
After you dial the exchange line access number, the external dial tone indicates that an external line is free. The external number can then be dialled.
You can also request an exchange line from the telephony app. Tap on to display the list of external contacts and dial an external account (public exchange). Tap on to display the list of internal contacts. Only internal telephone numbers can be dialled.
The exchange line access number of the PBX is „0“ in the factory settings but can be changed in some PBXs.
 
Enter the exchange line access number. When used as a system telephone the exchange line access number is automatically transferred from the PBX.
 
Keypad sequence 
Keypad sequences enable you to control features by entering characters and number sequences via the phone keys, e.g. to make pick-ups or InterCom announcements.
Important: If operated as a system telephone on the PBX, the keypad sequences are predefined by the PBX. The PBX overwrites any keypad sequences you have entered manually.
Note: When selecting the type Public exchange (Internet VoIP provider) under Settings > Providers & PBXs > Configure > Provider > Type some keypad sequencens and the exchange line access number are not available.
 
Enter keypad sequence  : The following keypad sequences for PBX operation are available:
Pick-up: The keypad sequence is required to carry out pick-ups on PBXs where a pick-up sequence is mandatory for this function. Information from SIP notify messages are not taken into consideration.
InterCom announcement: The keypad sequence is required to make an InterCom announcement. InterCom announcements can also be made on accounts where the telephone is not used as system telephone.
InterCom hands-free calling: The keypad sequence is required to enable Intercom hands-free calling at a destination which does not support automatic call acceptance.
Alarm confirmation: The keypad sequence is required to acknowledge an alarm call with the keypad.
Waiting loop: For moving into the waiting loop.
 
Evaluation of incoming SIP messages
Depending on the SIP provider, the phone number of the caller can be in different locations in the SIP message. The following options and settings must be selected and made to determine at which location of the SIP message the phone number shall be searched.
Two different phone numbers can be transmitted: The actual phone number of the caller (screened number/network provided number) and a number self-selected by the caller (unscreened number/user provided number). If a network provided number is extracted from the SIP message, functions such as Call Through can be used.
Evaluation type 
Standard: Selects an evaluation method for phone numbers which will work for most providers.
Standard (use From headers as screened-number): Selects an evaluation method for phone numbers which will work for most providers. The user provided number (corresponds to CLIP no screening) of the header “From“ will be taken over into the network provided number. The phone numbers can, for example, be used for Call Through via VoIP.
Caution: If you have selected this option, your telephone and your PBX are vulnerable to attacks, for example by transmission of a phone number manipulated by the attacker. Therefore, to protect your telephone and PBX, assure yourself that the VoIP provider has integrated sufficient security measures in the number presentation.
As described in RFC3325: Selects a phone number evaluation method in accordance with RFC 3325 (http://www.ietf.org/RFC/rfc3325.txt). This method of evaluating phone numbers is used if the number presentation is guaranteed by security mechanisms.
User-defined: If the Default and As described in RFC3325 options are unsuccessful, you can also define your own method of evaluating a phone number within certain limits.
Notes:
The phone number to be evaluated should preferably be canonical (+445306.... or 00445306…). Otherwise it is not used for evaluation.
The designations used for the settings match the designations used in Wireshark.
Network-provided number: Network-verified incoming phone number: The sequence entered in the invite is searched for the phone number which can be analysed.
User-provided number: Non-verified incoming phone number (corresponds to CLIP no screening): The sequence entered in the invite is searched for the phone number which can be analysed. Optional: is not transferred by all providers.
User-provided name: Caller's name in plain text. Optional: is not transferred by all providers.
 
Internationalise unknown numbers  
 Phone numbers which are not transferred in the usual format: (e. g. +445306… or 00445306…) are converted into this format.
 
 
Number presentation (outgoing)
You need to input the following information to ensure that the telephone transfers the data (e.g. the called phone number, its own phone number) to the provider in the correct format:
Format of called party number  : Format, in which the provider requires the phone number of the called party number to transfer the call.
Note: The telephone automatically converts called phone numbers into this format when it makes calls via the provider. If you dial a number without a prefix, the local area code therefore applies (Settings > Accounts > Country and area code).
With country and area code (0044 5306): The called phone number is sent to the provider along with its entire country code (e.g. 0044).
With country code and area code (+44 5306): The called phone number is sent to the provider with a plus sign (+) and the entire country code (e.g. +44).
With country code and area code (44 5306): The called phone number is sent to the provider with a shortened country code (without international prefix (number), e.g. 44).
With area code (05306): The called phone number is sent to the provider without a country code.
Unchanged (as dialled): The called phone number is sent to the provider unchanged.
Format of own number  : Format in which your own phone number, that is to be transmitted, is to be supplied to the provider.
Note: Your own phone number is automatically converted into the selected format. If you have entered your own phone numbers in the telephone without prefixes, the local area code applies to them during conversion (Settings > Accounts > Country and area code).
With country and area code (0044 5306): Your own phone number is sent to the provider along with its entire country code (e.g. 0044).
With country code and area code (+44 5306): Your own phone number is sent to the provider with a plus sign (+) and the entire country code (e.g. +44).
With country code and area code (44 5306): Your own phone number is sent to the provider with a shortened country code (without international prefix (number), e.g. 44).
With area code (05306): Your own phone number is sent to the provider without a country code.
Only port number (MSN / DDI + main number): Only your own connection number (MSN/DDI main number + DDI) is to sent to the provider.
dial (DDI): Only your own Direct Dial In (DDI) is sent to the provider.
Method for calling line identity restriction  : Calling line identity restriction by the provider with or without display text.
No display text: If calling line identity restriction is enabled, no display text is sent to the called phone number by the provider
Anonymous: If calling line identity restriction is enabled, the display text "anonymous" is sent to the called phone number by the provider.
User Anonymous: If calling line identity restriction is enabled, the display text "user anonymous" is sent to the called phone number by the provider.
Manner of number presentation  : Range in which the provider expects to receive phone numbers from the telephone.
In the display text: Your own external number, which is to be transmitted to the called phone number, is transferred to the provider in the "Display text" field.
In the username: Your own external number, which is to be transmitted to the called phone number, is transferred to the provider in the "Username" field.
As described in RFC3325 with P-Asserted-Identity: Your own external number, which is to be transmitted to the called phone number, is transferred to the provider in the "P-Asserted-Identity" field.
As described in RFC3325 with P-Preferred-Identity: Your own external number, which is to be transmitted to the called phone number, is transferred to the provider in the "P-Preferred-Identity" field.
Custom: Uses user-defined settings for number presentation.
Notes: The Custom option requires additional settings:
From
Settings for setting up the "From" header.
P-Asserted-Identity
Settings for setting up the P-Asserted-Identity" header.
P-Preferred-Identity
Settings for setting up the "P-Preferred-Identity" header.
 
 
Alert-info
Call type determination via alert-info 
A specific ringtone can be assigned to the different call types of a SIP account. The required information is transmitted by the SIP provider.
Requirements:
To assign a ringtone to a call, the link leading to the ringtone must be set into the alert info header.
<http://my.server.com/my_ringtone.mp3>
Subsequently, the link is evaluated in two different ways:
Note: Some providers already have fixed information for some call types.
An alert info can be entered for the following call types:
Internal calls 
Internal priority calls 
External calls 
Group calls 
Alarm calls 
Wake-up calls 
Door calls 
Silent calls 
User-defined call type (up to 5 different call types) 
Likewise, images belonging to a call can be transmitted via SIP. For this, the link to the image must be set into the call info header and the parameter "purpose" must be set on "icon".
<http://my.server.com/my_image.jpg>;purpose=icon>
This will also work for webcam images. Additionally, it can be entered in which time intervals the image shall be reloaded. For this, the parameter "refresh" is used. The entered value equals the repetition interval in milliseconds.
<http://my.server.com/my_webcam.jpg>;purpose=icon;refresh=333>
Webcams can also be used without entering a value for this parameter. After 3 and 10 seconds, the phone reloads the image. If it has changed during this time interval, subsequently, it is reloaded regularly. If this is not desired, 0 must be entered for the "refresh" parameter.
<http://my.server.com/my_webcam.jpg;purpose=icon;refresh=0>

COMfortel 1400 IP/2600 IP/3600 IP - Firmware V2.8 - Advanced Information V09 03/2024