Automated Provisioning
To avoid mis–configuration and reduce complexity, we highly recommend our integrated autoprovisioning system, configurable at https://sip.vetta.net.nz/pbx/phones.php.
Manual Configuration
Username: Use the username from your SIP Extension (SIP Peer)
Password: Use the password from your SIP Extension (SIP Peer)
Auth ID: Use the username from your SIP Extension (SIP Peer)
SIP Server 1: sip.vetta.net.nz
SIP Server 2: sip.vetta.net.nz
Transport: DNS NAPTR, NAPTR, DNS SRV, SRV, TCP, UDP (use the first option in the list that is available in your SIP client)
For TCP or UDP, use port 5060.
For Voice Engineers
Keepalive / OPTIONS
By default, we require SIP OPTIONS, preferably configured to poll every 30 seconds. We
also support SIP NOTIFY and RTP keepalive, which are disabled by default.
DTMF
For DTMF, we prefer RFC 2833 compliance.
Outbound Proxy
We recommend only configuring outbound proxy if it’s needed, as it can interfere with
failover / failback redundancy.
Outbound Proxy 1: proxy1a.sip.vetta.net.nz
Outbound Proxy 2: proxy1b.sip.vetta.net.nz
Supported Codecs
• OPUS
• iLBC
• G.729A
• G.722
• G.711 alaw
• G.711 ulaw
• GSM
Fax
We recommend using G.711 alaw primarily, with G.711 ulaw as backup.
Common Problems
One-way audio
If you’re experiencing a scenario where you only hear audio in one direction (either the caller only hearing you, and you can’t hear them, or vise versa), please try the following. Note this scenario is most common when behind Carrier-NAT, or when your SIP device is behind multiple layers of NAT.
- In your router(s), ensure SIP ALG / SIP NAT / SIP Helper (this setting name differs in different vendors) setting is disabled.
- In your Cloud PBX, navigate to the affected extension and set the “NAT” setting to “Yes”.
- Restart/re-register your SIP device (e.g. softphone or IP phone)
Comments are closed.