[email protected] wrote: > 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 >
Does your asterisk server have two network interfaces, one with a private IP address and another one with the public one? Did you try adding "canreinvite=no" to your 101 friend sip entry? What does the SIP debug say? -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com _______________________________________________ -- 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
