We're now getting this problem on outgoing calls. I've forced the port to 100FD but still no joy. Anyone any ideas how to debug this- have added verbose to logger.conf
Thanks for any help John 2010/1/4 John Taylor <j...@vetsurgeon.org.uk>: > I have recently moved our asterisk server from our LAN to a Debian > Lenny server (Asterisk 1.4.21.2~dfsg-3 ) with a public IP outside our > network. Our phones are behind a natted firewall. An ITSP provides a > PSTN to SIP termination for incoming calls > > Public ITSP -->Asterisk server-->Natted firewall-->extension (192.168.1.x) > > Everything works fine (incoming/outgoing audio etc.) except > occasionally an incoming caller is cut off whilst the called extension > stays in the call and can hear a DTMF tone (multimon recognises it as > tone "D"). The asterisk log file shows the call stays active despite > the incoming caller being cut off. This has happened to all our > extensions at some point (a combination of Snoms and Funkwerks). It > happens fairly infrequently, and can happen at any point during a > call. > > The public Lenny server's asterisk config is exactly the same as our > LAN Ubuntu asterisk server where we never had this problem. The only > difference is that the ITSP trunk is now ulaw rather than ilbc. > > Can anyone help? Relevant files below (trunk and extension codecs are both > ulaw) > > John > > > example extension in sip.conf: > [203] > type=friend > username=203 > secret=xxxxxx > host=dynamic > dtmfmode=inband > call-limit=2 > qualify=yes > nat=yes > > > /var/log/asterisk/messages: > [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing > [301xx...@fromvoipfone:1] Set("SIP/301xxxxx-09f74a00", "oh=0") in new > stack > [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing > [301xx...@fromvoipfone:2] NoOp("SIP/301xxxxx-09f74a00", "01295259352") > in new stack > [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing > [301xx...@fromvoipfone:3] GotoIf("SIP/301xxxxx-09f74a00", > "0?bankhols|200|1") in new stack > [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing > [301xx...@fromvoipfone:4] GotoIfTime("SIP/301xxxxx-09f74a00", > "08:30-18:00|mon-fri|*|*?day|100|1") in new stack > [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Goto (day,100,1) > [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing > [...@day:1] AGI("SIP/301xxxxx-09f74a00", "/home/john/phpagi/lookup") > in new stack > [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Launched AGI Script > /home/john/phpagi/lookup > [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- AGI Script > /home/john/phpagi/lookup completed, returning 0 > [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing > [...@day:2] Set("SIP/301xxxxx-09f74a00", "CALLERID(name)=xxxx") in new > stack > [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing > [...@day:3] Macro("SIP/301xxxxx-09f74a00", "monitor|01327xxxxxx|"in"") > in new stack > [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing > [...@macro-monitor:1] Set("SIP/301xxxxx-09f74a00", > "CALLFILENAME=/home/john/asterisk/asterisk_recordings/"in"-20100104_095856-01295259352") > in new stack > [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing > [...@macro-monitor:2] Monitor("SIP/301xxxxx-09f74a00", > "wav|/home/john/asterisk/asterisk_recordings/in-20100104_095856-01295259352|m") > in new stack > [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing > [...@day:4] Dial("SIP/301xxxxx-09f74a00", > "SIP/203&SIP/204&SIP/206&SIP/207&SIP/220&SIP/221|20|t") in new stack > [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Called 203 > [Jan 4 09:58:56] WARNING[10712] app_dial.c: Unable to create channel > of type 'SIP' (cause 3 - No route to destination) > [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Called 206 > [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Called 207 > [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Called 220 > [Jan 4 09:58:56] WARNING[10712] app_dial.c: Unable to create channel > of type 'SIP' (cause 3 - No route to destination) > [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- SIP/220-09fe7748 is ringing > [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- SIP/206-0a005eb8 is ringing > [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- SIP/207-09fe2c98 is ringing > [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- SIP/203-0a001138 is ringing > [Jan 4 09:58:57] VERBOSE[10712] logger.c: -- SIP/220-09fe7748 is ringing > [Jan 4 09:58:57] VERBOSE[10712] logger.c: -- SIP/203-0a001138 is ringing > [Jan 4 09:58:58] VERBOSE[10712] logger.c: -- SIP/220-09fe7748 is ringing > [Jan 4 09:58:58] VERBOSE[10712] logger.c: -- SIP/203-0a001138 is ringing > [Jan 4 09:58:59] VERBOSE[10712] logger.c: -- SIP/203-0a001138 > answered SIP/301xxxxx-09f74a00 > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users