I have found a SIP provider that i can use with voicent auto dialer, however while installing sip information, we recieved an error message SIP 488 error. The tech informed me it was due to a codec problem. The codecs that the sip provider uses is G711u which is not part of the codes available from your list.

I was asked to find out from you if there is anyway of adding that codec or if with the full version of the program you offer that specific codec.

Please advise as i am ready to purchase program however need to make sure SIP provider can work with Voicent. Please let me know asap as i will need to order from both voicent and sip provider right away as i will need to use the autodialer.

The G711u codec is named PCMU codec. And it is selected by default. For more information, please see Setup SIP Codec

You should use the shareware edition to test. If it does not work, it won’t work for the paid version.

STUN, Simple Traversal of UDP through NATs , is a protocol used to work around the problems caused by router/firewall for VOIP applications. It is also known as Session Traversal Utilities for NAT.

The basic problem of a router/firewall for VOIP traffic is that a computer’s IP address inside NAT cannot be seen from outside. So in order for other application to send audio strean to a VOIP application inside a router, it must know the outside IP address used by the router. And in order to discover the outside IP address, a STUN client sends a request to a STUN server outside the router/firewall. Since the request has to pass through the router, it will carry the outside IP address in the request. The STUN server simply returns the outside IP address to the STUN client.

To setup Voicent software, if your computer has to pass through a router in order to reach your VOIP service, then you must specify a STUN server. If your computer and the VOIP service or PBX system are on the same local area network (LAN), then DO NOT specify a STUN server (if you do, the audio will be sent ourside the router and you will not be able to hear anything).

Starting from release 7.2.3, you can select which SIP codecs to use from a set of existing ones. The reason for that is some VOIP service or PBX system limit the size of SIP message. So if the SIP message is too large because the codecs information is too long, the SIP message is dropped.

Normally, you can use the default selections, which include gsm, pcma, pcmu, and ilbc codecs. But if your VOIP service or PBX require or use different codecs, such as G.722, then you can set that value from Voicent Gateway > Setup > options > SIP > Codecs window.