Trunks are the equivalent of lines to your carrier. The trunk menu allows you to instruct the PBX what to expect from your telco and what to send to your telco. It is about number formats and protocols used. Make sure you have received all details from your SIP provider to make configuration of your trunks simple.

You can add or edit trunks in Communication/Trunks.

General SIP settings

• Name: text field used as reference in the PBX. Allowed characters aA-zZ, 0-9, _ and –
• IP/hostname: the hostname of SIP proxy of your SIP provider
• Trunk type:
  • external (default): meaning this is where external (e.g. PSTN) calls come from and can be sent to. This sets the right security measures in place to prevent abuse.
  • internal: to connect one or more PBX‘s together or gsmbox’s etc. Setting the type to internal means internal numbers and services can be reached directly.
• Port: the TCP/UDP port to communicate on. Default value 5060
• Authentication: How to authenticate your PBX at your SIP provider
  • None: this assumes you provider uses your IP for authentication or really no authentication at all
  • Password (default): this will make the fields username and password editable
• Our username: the username the SIP provider gave you to identify your system
• Our password: the password belonging to “Our username”
• Peer’s authentication: sets how you want your SIP provider or some other PBX authenticate itself to your PBX
  • IP (default): the IP from hostname or the IP from register hostname will be accepted. It is still recommended to close up your PBX with the built-in Firewall and Intrusion detection system
  • Password: demand username/password authentication from your other PBX, SIP provider or GSM box etc.
• Peer addresses: SIP trunk provider (proxy/sbc) IP. If multiple, then comma seperated. This is a required field if “Peer authentication” is set to IP.
• Peer’s username: the username of the peer if authentication is set to password
• Peer’s password: the password belonging to the peer username
• SIP transport: The (layer 2) protocol to be used with the Peer. Default is UDP as this is the most efficient for VoIP. Other values are: UDP/TCP, TCP/UDP or TCP.
• Fall back trunk: if you have multiple either SIP or ISDN trunks you can use those automatically if this trunk fails. The end-user would not notice. This is a very common solution with standalone units (AOX on site PBX) where ISDN is available.
• Outbound proxy: Optional outbound proxy for your SIP trunk.  Hostname or IPv4/IPv6 address, with optional port (e.g.: ‘sip.example.org’, ‘192.168.1.1’, ‘sip.example.org:5070’, ‘1234:5678::1000’, ‘[1234:5678::1000]:5070′).

Registering your PBX with SIP to your provider is very common. It it used to tell your SIP provider the IP address and service port your PBX can be reached on for inbound traffic. It has nothing to do with the actual authentication, although often the same username and password is applied. The PBX supports exceptions to these rules. It is recommended to start with the default unless instructed otherwise by your SIP provider.

• Register: tick the checkbox to enable.
• Register refresh: time in seconds to refresh the register, 600 seconds is a good default. If you have more than 1 SIP trunk defined, only one trunk can set the register refresh value. All other trunks must use the same value. You can choose yourself which sets the value, by default the first trunk created, sets the value.
• Registration username: default value is the same as the username in tab General. Only change if instructed by your SIP provider
• Authentication username: The register allows for a separate username for authentication. This is very rare. Do not use unless specifically instructed.
• Registration domain: default value is the same as the IP address/hostname in tab General. Only change if instructed by your SIP provider.
• Registration host: a hostname or Ipv4/IPv6 address with optional port number can be entered there. Only change if instructed by your SIP provider.
• Local DID: Some SIP provider require to register the phone number upon SIP register. The values you can choose from are based on the entries listed in the Dial plan/inbound menu, so you first need to create those. This field is sometimes used by SIP providers targeting the consumer market. Business focussed SIP providers rarely use this. Only use when specifically instructed.

Miscellaneous settings for the SIP communication with a SIP provider

• Caller ID passing: set which method to use to send Presentation (=CLIP) or anonymous (=CLIR).
  • None: the SIP “From:” header is used by your SIP provider. This is the default setting that works with most SIP providers
  • P-Preferred-Identity: the next most used SIP method for send CLIP/CLIR information
  • Remote-Party-Id: an old SIP method not commonly in use, is here for legacy systems
  • Both: Use P-Preferred-Indentity and Remote-Party-Id. The use of “From:” is always possible regardless which method selected
• From username: what value to set in the SIP “From:” header. Use the default value “Caller-ID” unless you know what you are doing or are being in instructed to do so.
  • Caller-ID: (=default) use the number either set for callerID in trunk, IP phone, Phone user or your SIP provider sets it for you
  • Anonymous: force anonymous always. This may be refused by your SIP provider resulting in not being able to make calls. Use only when your SIP provider confirmed you can. There are other ways to set anonymous dialing for users (see outbound, Phones or user settings).
  • Custom: set some string to your own wishes. This may result in not being able to dial. Check with your SIP provider if this setting is needed at all. Most SIP providers do not need it.
• NAT support: add extra NAT information in the SIP headers. Default this is set to yes. May be switch off when not using NAT
• Keep alive: send SIP OPTIONS with some regular interval, to monitor the SIP end-points and/or to keep a NAT entry in the firewall open.
  • Default: every 60 second
  • 1, 10, 30: pre-set values
  • No: do not send SIP OPTIONS
• Reinvite: Allow SIP re-INVITE to be exchanged with the Provider. Few providers support this option. It is used with call transfers and with T.38 fax sending. Default value is off. If you suffer from one way audio you can try changing this setting.
• Account code: set some string that is written in the CDR data account code field. Default value the username for your SIP account

Set how to format the outgoing numbers to your SIP provider. The default values work for the vast majority of the SIP providers.

• Send normalized e.164 number: normalise according to E.164. Default value is ‘On’
• Prefix E.164 (with): what prefix to use. Most common is “00” or ”+” and sometimes just nothing
• Country code: default empty only needed when your SIP provider requires always an international format
• Area code: default empty, only needed when you want to be able to dial local numbers. You must change the dial plan then accordingly to recognise the number used for local numbers.
• Strip country code: only used when calls come in via SIP and are diverted via ISDN. Not commonly used
• Strip area code: only used when calls come in via SIP and are diverted via ISDN. Not commonly used
• Allow arbitrary outbound caller-id: This is an advanced function only necessary in very rare configurations. Disabled is the recommended and default setting. If you really need it, you’ll know how to use it. This can potentially cause issues as the upstream provider can reject the call due to a foreign caller id.
• Default outbound caller-id:  the default outbound caller-id used if not set in the outbound trunk, phone or user.

Only make changes here when you know what you are doing. Setting this incorrectly can result in poor sound quality and/or very limited number of concurrent calls. If not set properly, the PBX will do codec translation. This is a very expensive and CPU intense process that will affect the performance of your system. Make changes here with care and verify that your system is not transcoding.

The default value works just about always if the Axeos software is used in Europe. No need to make changes here.

When you set up trunks between some PBX‘s as internal trunks you may want to make changes here, though there is no need to for making things work.

g729 codec support ended with Axeos version 4.0. If you are using this codec in a 3.x pbx be aware that this codec will no longer work when you upgrade to 4.x. Also please be aware of the inherent disadvantages of g729: dual-tone multi-frequency signaling (DTMF), fax transmissions, and high-quality audio cannot be transported reliably using the g729 codec. The GSM codec can be a valid alternative choice for your low-bandwith codec requirements.