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Jezwalton

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Hi Guys,
I run a asterisk PBX on 10.0.0.154
I have KWF on 195.74.*.162
I have a second IP setup on the NIC of 165

I have two rules
Rule 1 Out
Source 10.0.0.154
Dest - Public Netconnection
Service ANY
Translation - NAT to 195.74.*.165

Rule2 In
Source IP's of Sip provider
Dest : 195.74.*.165
Service ANY
Translation MAP to 10.0.0.154

The PBX registers with the address 10.0.0.154 (Should be public address .165)
We can call out Fine IE inside firewall we can talk and outside you can hear.
However vice versa does not work ?

I did specify sip etc and removed the Protocol inspectors etc. Still no Joy.
I spoke to one of Kerio's Techs in the UK but he was struggling with it too..


Help please

Jez
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Johnvoip

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VoipSipSdk

I am now looking for voip solutions. And found information about Voip sdk.
According to their website www.voipsipsdk.com
Voip sdk is based on IETF standards (SIP, STUN, etc.), so it should be compatible with other standard based products such as Asterisk, OpenSER other.

They have all features I need:
# Dynamically loadable codecs
# Registrar support
# Play wav files into conversation
# Record conversation into file
# Hold/Retrieve call
# Forward Call (Blind Call Transfer)
# Transfer Call (Attended Transfer)
# Mute Sound
# VPN support
# Noise reduction
# Auto gain
# Jitter buffer parameters
# Samples on Delphi, C#, VB, VB.NET, C++ 2005, C++ 6.0, HTML (SIP ActiveX)
# Windowless samples on C++ and .NET
# DTMF
# Adaptive silence detection
# Adaptive jitter buffer
# STUN support
# Comes as ActiveX control

But before I will download the evaluation version I would like to hear other people experience.
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spikeinin

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You found information about Voip sdk when you are looking for voip solutions. Voip sdk is based on IETF standards (SIP, STUN, etc.) so it should be compatible with other standard based products such as Asterisk, OpenSER other VOIP..


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