What are the asterisk NAT settings in effect for each of the SIP phones? I'd be inclined to turn them both ON to ensure that symmetrical RTP in being used. Also make sure that canreinvite is OFF for both.
----- Original Message ----- From: "Ian Pattison" <[EMAIL PROTECTED]>
To: <[email protected]>
Sent: Thursday, April 07, 2005 4:49 AM
Subject: [Asterisk-Users] SIP - SIP Problems
Hi Everybody...
Continuing the litany of problems I'm experiencing with my new system I'm =etting issues connecting between SIP phones.
A bit of background... I have an asterisk server running in a central =ocation where I have two incoming analog lines connected to FXO ports, =wo analog phones connecting to FXS ports and a single SIP phone. In =ddition I have a remote site connected via a CIPE VPN (ok..ok I know it's =ot a real VPN...) with a single SIP phone.
Calls initiated from the remote SIP phone to the central SIP phone often =ave trouble... the user of the central phone cannot hear anything from =he remote phone although everything is heard at the remote phone. If the =emote phone calls either outside or to one of the Zap phones there is no =rouble. If the local SIP phone calls the remote SIP phone there is no =rouble. Both phones are from the same vendor, have the same firmware and =he same configuration with the exception of phone number, PIN, IP address =tc.
What am I doing wrong here?
Ian
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