[Asterisk-Users] SIP - SIP Problems

[Rod Bacon]

I'd personally be using Ethereal to look inside the SIP messages for the SDP 
info and checking the source/destination of the resultant RTP stream. 
One-way audio is typical of NAT issues. Although you are running a VPN (of 
sorts) I suspect that your SDP messages are getting screwed up somewhere.

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" <ianp at technologyassociates.ca>
To: <asterisk-users at lists.digium.com>
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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