Hi all, After running for months without issue I've got a situation where incoming SIP calls to my asterisk server are failing while outbound calls appear to be working as expected.
The server is a gateway between my home LAN and a broadband cable connection with a dynamic IP. The gateway runs FreeBSD 7.1 and Asterisk 1.6.0.15 (built from ports) and registers to my ISTP no problem. Outgoing calls can be made successfully and no error messages or warnings are reported by Asterisk. However, incoming calls appear to enter my dialplan as desired and go so far as to start ringing my SIP phone (Grandstream GXP-2000) but drop after two rings. The caller gets a busy tone and that's it. If I answer the call before the two rings I just get a moment of dead air and it drops in the same way. In the asterisk console (and log file) I see these messages at the fail point: [Oct 9 12:42:47] WARNING[1056]: chan_sip.c:2917 retrans_pkt: Maximum retries exceeded on transmission b734bd26-2fae-122d-91a5-653b331e358a for seqno 121440337 (Critical Response) -- See doc/sip-retransmit.txt. [Oct 9 12:42:47] WARNING[1056]: chan_sip.c:2944 retrans_pkt: Hanging up call b734bd26-2fae-122d-91a5-653b331e358a - no reply to our critical packet (see doc/sip-retransmit.txt) Okay, so I verified that my firewall is properly accepting traffic on the range of SIP and RTP ports as specified by my ITSP. After sending them a sip debug trace my provider said this: "It appears that your machine is not receiving replies when it tries to acknowledge the incoming call back to our server. This could be a firewall issue or potentially something else that changed without your knowledge." Furthermore, they suggested I might try registering and connecting directly to their Asterisk using only the Grandstream phone. I tried this and...surprise! Both inbound and outbound calls work fine but leave me without voicemail or any other services my PBX would be providing. Right, so now I'm thinking there must be something wrong with my Asterisk configuration yet I've made no config changes that would account for the sudden (and consistent) incoming call failures. Here's the relevant portions of my sip.conf if it helps (with credentials and ips replaced by Xs): [general] alwaysauthreject=yes dtmfmode=auto disallow=all allow=ulaw register => XXXX:[email protected]:5060 register => XXXX:[email protected]:5060 [101] type=friend context=websage host=dynamic deny=0.0.0.0/0 permit=XXX.XXX.XXX.XXX/24 qualify=yes secret=XXXX mailbox=...@default accountcode=101 I'm now at a complete loss for how to proceed trying to resolve this and hoping someone with more experience than I on the list might have some ideas or suggestions. Any and all advice is warmly appreciated. Cheers, GM -- Greg Maruszeczka _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
