To be able to make phone calls a SIP trunk should be registered on Ayrix. Additionally, DID numbers have to be defined in order to receive incoming calls.
Adding a new SIP Trunk #
To add a new SIP Trunk, click on “Add SIP Trunk” to open the Country/VoIP Provider window.
- Select the country where the VoIP provider operates
- Select the VoIP provider where the SIP Trunk that needs to be registered
- Click the “Add”-button
Now the “Basic Settings” window opens with the new created SIP Trunk.
Configuring the SIP Trunk #
In the “Basic Settings” tab some values of the VoIP provider are already set. To complete and save the SIP Trunk data you must add some additional information. A window with Configuration settings panel and Codecs settings panel opens.
- Fill in a Default Caller ID.
- Enter SIP User-ID and the password, according the providers Information.
- Click “Save”-button
The “Default Caller ID” may look like an inbound number, but in fact it is a recognition number (or office number) to identify your Trunk by your provider. In the tab DiDs inbound numbers are configured.
The next tabs are shown:
- Basic Settings
- SIP Options
- Inbound headers
- Outbound headers
This configuration windows contains settings to create and connect a SIP trunk.
|Configure SIP Trunk|
|Name||Give your trunk a recognizable name. for ex. “Trunk-to-Provider”|
|Registrar / Server||The registrar server of the chosen VoIP provider.|
|Outbound proxy||A proxy that receives requests from a client, even though it may not be the server resolved by the Request-URI.|
|Port||The port which the SIP signaling communication uses.|
|Default Caller ID||Default number displayed at an outbound call destination|
|Max simultaneous calls||Limit the maximum of simultaneous incoming and outgoing calls. “0” means maximum possible|
|Authentication||Authentication settings for your trunk|
|SIP User-ID||User-ID needed for authentication at the VoIP Provider|
|Password||Password needed for authentication at the VoIP Provider.|
|Format caller-called ID|
|Caller ID Regex, Called-ID Regex||Regex rule formatting for caller or called ID to display the called party in an outbound call a correct number to call back.|
The right panel shows the Codecs window. The codecs you want to apply for the SIP Trunk has to be selected from the tab “Available Codecs”. Various codecs can be selected from the list and moved in the selected Codec list.
The used codec is a grade of the speech quality. The use of a particular codec is negotiated by the SIP protocol due call setup. The first codec that matches on both sides is chosen. In the Selected Codecs you can rank the codecs. Selecting trunk codecs, enables also default codecs for extensions.
|Selected Codecs||List of the selected Codecs, that will be negotiated with the opposite party|
|Available Codecs||List of available Codecs in the Ayrix|
A Codec is the digital translation of the voice to transmit a media RTP or SRTP stream,
A description of the available codec usage.
|G711-law||64Kbit/s pcm coding, with a fixed 8Kbit sample rate narrowband audio codec standard, except USA and Japan.|
|G711-ulaw||64Kbit/s pcm coding with 8Kbit sample rate narrowband audio codec American and Japan|
|G722||G.722, 7 kHz wideband audio sampled in ADPCM 58K, 56K und 64Kbit/s format streaming|
|G726||G726 is an ADPCM speech codec with 2vairabele sample rate and multiple transfer rates of 16,24, and 40Kbit/sec|
|G729||G729 narrow-band vocoder-based audio data compression algorithm, with Coding of speech at 8 kbit/s|
|GSM||Global System for Mobile Communications (GSM) is a codec in cellular networks|
|LBC||(Internet) Low Bitrate Codec (iLBC) is a narrowband speech audio coding format with different data frame formats.|
|Siren7||Siren 7 (or Siren) provides 7 kHz audio, bit transfer rates 16, 24, 32 kbit/s with an audio sampling frequency 16 kHz.|
|Siren14||Siren 14 (or Siren14) provides 14 kHz audio, bit rates 24, 32, 48 kbit/s for mono, 48, 64, 96 kbit/s for stereo transfer rate with an audio sampling frequency 32 kHz.|
|Speex||Speex is an audio compression codec tuned for the reproduction of human speech. Speex is targeted at voice over IP (VoIP)|
|Speex16||Speex16 is an audio compressed codec using a fixed 16kbit audio sampling rate.|
Direct inbound dialing (DiDs).
To connect with the outside parties, Direct inbound dialing (DID) numbers have to be added to the Trunk. These DiDs are the numbers you can dial in to address the destination. Go to the “DIDs” tab. A window with a listed panel and a configuration panel opens.
Select in the configuration panel your single number or a number block where the callers can dial in. You can add a single Number or a Number block.
Your new number block is now listed and available for assignment.
In this window can be set the sip options for the Trunk.
|SRTP||Enables signaling and media encryption on the trunk line (TLS & SRTP).|
|Language||Set Language to select country specific media, for ex. ringtones|
|Re-Register Timeout||Trunk guarding, Re-Register time-out in minutes.|
*v.0.21.1. SRTP is not yet activated. The function will be enabled in an upcoming release.
Due to the different RFC specifications, many signaling options for incoming SIP trunk lines remain. These SIP elements are inbound headers. Headers, or SIP elements that differs from the standard are known by your provider. For each SIP Message element, you can adapt several other SIP identification elements. On the webpage several inbound elements are listed.
We strongly advice to leave the values to default unless a specialist from your provider advice different.
|Call source identification|
|“CallerNum” caller’ number (default: From > user):||[A]default: From > user:|
|“OutboundLineId” Outbound Line caller ID taken from Outbound caller ID setting in managed console.||[A]Leave default value|
|*OutboundCallerid” Outbound caller ID taken from Extension setting in management console||[A]Leave default value|
|“CallerDispName” Display name of a caller as it is From Header – Provided by phone settings||[A]Leave default value|
|“CalledName” that has been dialed (Default To>display name)||[A]Default To>display name|
|“CalledNum” number that has been dialed (Default To>User)||[A]Default To>User|
|“CallerName” caller’s name (default from>display name)||[A]default from>display name|
|OriginatorCallerID Original Caller number will be sent||[A]Leave default value|
|“DevHostPort” source address/port of message||[A]Leave default value|
|ContactUrl usually content of the contact field||[A]Leave default value|
Selectable inbound elements [A]
|Contact: User Part|
|From: Display Name|
|From User Part|
|P-asserted Identity: Display Name|
|P-asserted Identity: User Part|
|P-Called Identity: Display Name|
|P-Called Identity: User Part|
|P-preferred Identity: Display Name|
|P-preferred Identity: User Part|
|Remote Party ID – called Party: Display Name|
|Remote Party ID – called Party: User Part|
|Remote Party ID – calling Party: Display Name|
|Remote Party ID – calling Party: User Party|
|Request Line URI: User Part|
|To: Display Name|
|To: User Part|
A variety of outgoing SIP trunk signaling options following different RFCs remains possible. These call source elements depend on specific country and/or provider settings. For each SIP Message element, you can adapt several SIP identification elements. On the webpage several outbound elements are listed that can be changed.
We strongly advice to leave the values to default unless a specialist from your provider advice different. Depending on the type of SIP element there are 2 selectable lists [B&C]
|Contact User Part||[B] Leave default value|
|Contact Host Part||[C] Leave default value|
|Remote Party ID -Called Party Display||[B] Leave default value|
|Remote Party ID- Called User Part||[B] Leave default value|
|Remote Party ID – Called Party Host Part||[C] Leave default value|
|Remote Party ID – Calling party Display Name||[B] Leave default value|
|Remote Party ID – Calling Party User Part||[B]Leave default value|
|Remote Party ID – Calling Party Host Part||[C]Leave default value|
|P-Asserted identity Display Part||[B]Leave default value|
|P-Asserted identity User Part||[B]Leave default value|
|P-Asserted identity Host Part||[B]Leave default value|
|P-Preferred-Party-ID User Part||[B]Leave default value|
|P-Preferred-Party-ID Host Part||[C]Leave default value|
|P-Called-Party-ID Display Name||[B]Leave default value|
|P-Called-Party-ID User Part||[B]Leave default value|
|P-Called-Party-ID Host Part||[C]Leave default value|
[B] Selectable outbound elements (self-explaining)
|CallerNum number that has been dialed (default: To > User)|
|CallerDispName Display name of a caller as it is in From Header|
|CallerNum caller`s number (default From->user)|
|LineID internal number of lines|
|LineNumber external number of lines|
|OriginatorCallerID original caller number will be sent|
|OutboundCallerID outbound caller Id taken from Extension settings in management console|
|OutboundLineID outbound Line caller ID taken from outbound caller ID setting in management console|
|Leave default value|
[C] Selectable outbound elements (self-explaining)
|ContactUri usually content of Contact field|
|DevHostPort, source address/port of message|
|GWHostPort, gateway/provider host/port|
|OutHostPort: outbound proxy host/port|
|Leave default value|
Housekeeping: see chapter 18.2 SIP provider/country dependent behavior…)
Trunk Status #
The Trunk list panel, the STATUS field shows you the actual trunk status of the configured trunk(s).
|Connection established (OK)||REGISTERED_[DAY*]|
|Incorrect URL Registrar||Request sent|
|Incorrect port||Request sent|
|Not active or no establishment (other failures)||Unknown|
*Day of the week, MON, TUE, WED, THU, FRI, SAT, SUN
Registration of a trunk may take up a few minutes. Be patient, if the lamp does not immediately switch from Registering into Registered_[DAY] Refresh your website!