Removed NAT from both phones but left "canreinvite=no" for both. Looking better.
>>> [EMAIL PROTECTED] 07/04/2005 09:23 >>> Don't ask me why but I've got some limited connectivity now... I initially disabled "canreinvite" and enabled NAT on both phones. They connected to * just fine but when I attempted to make calls I received no audio on the remote phone... can't say for sure on the local one. I then disabled NAT on the remote phone only and was able to make a 2-way voice call to the local phone (although it did take about 2 seconds for audio to kick in...) from my understanding of the situation this config should not work at all. Packet decodes are my next step... has anyone here ever successfully had Ethereal running in text-mode only? My * box does not have X installed and is only accessible via SSH. Thanks, Ian >>> "Rod Bacon" <[EMAIL PROTECTED]> 06/04/2005 20:59 >>> 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" <[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 _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
