This guide introduces SIP registration with Yeastar VoIP PBX.
- Understanding SIP registration basic
- Troubleshoot extension registration issues
SIP Extension Registration
Registration associates user’s identification, or Address of Record (AOR), with one or more locations. It is simply a mechanism where a phone communicates "Hey, I'm Bob's phone... here's my username and password. Oh, and if you get any calls for me, I'm at this particular IP address." Here is the SIP registration flow:
Here is an example of SIP extension registration packet:
REGISTER sip:192.168.9.208:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.9.39:5065;branch=z9hG4bK68669670
From: "3009" <sip:email@example.com>;tag=1569518026
To: "3009" sip:firstname.lastname@example.org
CSeq: 1 REGISTER
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
User-Agent: Yealink SIP-T28P 220.127.116.11
Get the username and IP address of registration source
Get the username and IP address of registration destination
This field contains the URL at which the UA would like to receive request.
The authentication name of the endpoint.
An integral number of seconds, measured from the receipt of the request.
In the following article, we would introduce the configuration related to SIP registration.
- Concurrent Registrations: You can specify how many devices are allowed to register for this extension here. The maximum is 5.
- Register Name: Digest username parameter of Authorization field in the SIP REGISTER packet. For example:
Authorization: Digest username="3009", realm="YSAsterisk"
- Caller ID: username parameter of "From" URL
- Caller ID name: display parameter of "From" URL. For example:
From: "display3009" sip:email@example.com
- Register Remotely：When endpoint is locate in a different network segment with PBX, Register Remotely is needed for register.
- Transport: UDP, TCP, TLS
- IP Restriction: This option is for specific IP address register the extension.
- User Agent: When registering, package sent by SIP phones will contain the User Agent string. If the string doesn’t match the value here, registration will fail.
UDP Port: The UDP Port is used to monitor the port flux for PBX, the default value is 5060. Allow UDP port 5060 traffic received from other device.
TCP Port: Port traffic of TCP will also be allowed to access and registration. The default value is 5060.
Local SIP Port: The port that flows out the local will be random port range from 5062 to 5082.
Registration Timers: The registration duration for SIP extension, the default is 3600.
TLS: Encrypt data during data transmission. The default value is 5061.
*Note: Usage scenario of Local SIP Port
1) Sometimes the registration from PBX to provider is too frequent. The PBX with default SIP port 5060 will be blocked by the firewall or router of provider. To prevent this problem, our PBX support random port in registration after enabling the Local SIP Port option.
2) When there are multiple trunks on your device, provider/ end device can’t distinguish the traffic from same port. This option could help them use different port to distinguish the registration flow.
Linkus Server Settings
- Linkus Local Port: It is an login port for Local Linkus client and default is 8111.
- Linkus External Port: The external port for connection with remote Linkus client.
- SIP Remote Registration Port: The registration port for remote Linkus client to register. We need to enter the external port which mapped for SIP port 5060. The default is 5060 here.
*Note: 8111 is the port for Linkus Client to login and other data can also pass through the port.
Estou configurando o Fanvil X1P e o uso o PBX da yeastar S50. Embora os versos disponíveis sejam 2,8 ... superiores que sejam compativeis com PBX em referência. o problema eh que não consegue registrar nenhum telefone FANVIL X1P, tempo limite / tempo limite eh uma resposta que vem com status. Tentei Fanvil X3G e não tem nenhum problema. sera o problema do PBX S50? ou mesmo FANVIL X1P?