Hi all,
I'm trying to troubleshoot an issue with my SIP service. All outgoing calls work normally. The following is a SIP debug log from Asterisk. The test setup is as follows: One Yealink SIP-T22P phone (10.0.0.107), extension 10, configured to talk to my local FreePBX/Asterisk 11.0 server which is at 10.0.0.17. The Yealink phone doesn't seem to have any problem placing outgoing calls through the Asterisk server, which is registered to Diamondcard. I can reach both the Asterisk server itself (for example to use voicemail) or call any number on the PSTN. Likewise I have the server configured to pass incoming DID calls for myDIDnumber to extension 10. Calls from the PSTN to myDIDnumber ring the phone, including CID passing, and will connect a full duplex audio call session. The problem is that the phone won't stay connected longer than 13 to 17 seconds. When the phone is manually configured to use my account and password on the diamondcard servers directly, both incoming and outgoing calls work normally, with RTP/UDP port 5060 traffic passing through my NAT without trouble. I have made no special modifications to the NAT. 13 seconds after picking up an incoming call, the phone disconnects at the same time as the log shows this: [2012-11-11 01:51:40] WARNING[3201] chan_sip.c: Retransmission timeout reached on transmission [email protected] for seqno 103 (Critical Response) -- Seehttps://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 17853ms with no response [2012-11-11 01:51:40] WARNING[3201] chan_sip.c: Hanging up [email protected] - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). *The full log and configuration is at:* *http://pastebin.com/1Mgn72vN*
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
