Adding a SIP phone profile on the Teles gateway

Here we will try to register a phone on the Teles Gateway. First we need to create a SIP user profile in the “route.cfg” file. The VOIP profile code must be between the mapping operations and the VOIP profiles, so the other VOIP profiles will not influence your experiment, and you will not influence the others routing rules. After ;---VoIP Profiles of allowed SIP servers---, insert two commented lines “;---beginning of my tests (VOIP Profiles)---” and “;---end of my tests (VOIP Profiles)---”. Write all your VOIP profiles between these two lines. Your code must be then deleted and the original version must be re-uploaded as soon as you finish your experiments.

VoipPeerAddress will specify that this profile is used only for users coming from a specified IP Address. VoipOwnUser, and VoipOwnPwd specify user, access, and password for the SIP phone to be connected to Teles gateway. VoipSignalling=1 signifies SIP protocol (and not H.323). VoipAutoRtpAddr=Yes is important if you are behind a NAT. More complete list of parameters is shown on p.81 of the gateway manual [pdf].

Save and upload this profile to the gateway and start configuring the phone.

The Siemens C450IP SIP phone must be configured as shown in the screenshot below. Domain, proxy server address, and registrar server must all point to the Teles gateway. If our phone is behind NAT, we must use a STUN server. Teles gateway certainly has settings permitting a NAT traversal, but with this sample, a STUN server is necessary. We use our public STUN server us1.youroute.net (66.234.138.73) for this purpose:

Your phone must register on Teles, after these settings are in force.

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