Hello list ! I have the following problem at a customer :
Their is a firewall in between the internal network (with IP-phones) and the public Asterisk-server. I see the following message when "sip debug" enabled : [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] --- (11 headers 11 lines) --- [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Found RTP audio format 8 [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Found RTP audio format 101 [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Peer audio RTP is at port 192.168.0.24:11772 [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Found audio description format PCMA for ID 8 [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Found audio description format telephone-event for ID 101 alaw) d - 0x1 (telephone-event) [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] Peer audio RTP is at port 192.168.0.24:11772 [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] list_route: hop: <sip:ic...@192.168.0.24:5062> [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] set_destination: Parsing <sip:ic...@192.168.0.24:5062> for address/port to send to [Mar 24 11:19:36] VERBOSE[5087] logger.c: [Mar 24 11:19:36] set_destination: set destination to 192.168.0.24, port 5062 But when opening a range of ports on the firewall 11700 --> 11800, the audio is not coming through !! When opening the ports 11000 --> 11800, then the audio is coming through fine ! Can someone explain me why range 1 is not enough fot the RTP-traffic ?! Jonas.
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