[Asterisk-Users] SIP - SIP Problems

[Joseph]

On Wed, 2005-04-06 at 14:49 -0400, Ian Pattison wrote:
> Hi Everybody...
> 
> Continuing the litany of problems I'm experiencing with my new system I'm getting issues connecting between SIP phones.
> 
> A bit of background... I have an asterisk server running in a central location where I have two incoming analog lines connected to FXO ports, two analog phones connecting to FXS ports and a single SIP phone. In addition I have a remote site connected via a CIPE VPN (ok..ok I know it's not a real VPN...) with a single SIP phone. 
> 
> Calls initiated from the remote SIP phone to the central SIP phone often have trouble... the user of the central phone cannot hear anything from the remote phone although everything is heard at the remote phone. If the remote phone calls either outside or to one of the Zap phones there is no trouble. If the local SIP phone calls the remote SIP phone there is no trouble. Both phones are from the same vendor, have the same firmware and the same configuration with the exception of phone number, PIN, IP address etc.
> 
> What am I doing wrong here?
> 
> Ian

What *-ver. are you using?
I had a problem with SIP in ver.1.0.7 and 1.0.6 so I'm still on 1.0.5

-- 
#Joseph