kjcsb
2007-Feb-02 10:06 UTC
[asterisk-users] No RTP packets received by Asterisk when calling SIP to SIP
I have the following setup: UA1 (SPA2000) -- Nat1 -- Asterisk (public internet) -- Nat 1 -- UA2 (X-Lite) Relevant parts of sip.conf [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) externip = 60.234.100.100 ;External IP address localnet = 192.168.1.0/255.255.255.0 ;Local network address allow=all [1590] username=1590 type=friend secret=secret qualify=no port=5060 nat=yes mailbox=1590@device host=dynamic dtmfmode=rfc2833 context=test canreinvite=no allow=all [1593] username=1593 type=friend secret=secret qualify=no port=5060 nat=yes mailbox=1593@device host=dynamic dtmfmode=rfc2833 context=test canreinvite=no allow=all I have enabled rtp debugging and notice that Asterisk is receiving no rtp traffic. When I call from either UA to voicemail for example I see RTP traffic e.g. call from 1590 Got RTP packet from 60.234.200.200:38510 (type 0, seq 1245, ts 207620, len 160) Sent RTP packet to 60.234.200.200:38510 (type 0, seq 61963, ts 34880, len 160) e.g. call from 1593 Got RTP packet from 60.234.200.200:16470 (type 0, seq 892, ts 316685167, len 240) Sent RTP packet to 60.234.200.200:16470 (type 0, seq 1156, ts 15360, len 160) I thought that with canreinvite=no all audio would go through Asterisk. What have I missed? Asterisk 1.2.13 Fedora Core 5 Regards Cameron ___________________________________________________________ Copy addresses and emails from any email account to Yahoo! Mail - quick, easy and free. http://uk.docs.yahoo.com/trueswitch2.html -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070202/9e8e1f90/attachment.htm