[asterisk-users] caller getting cut off intermittently

[John Taylor]

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 <john at 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
> [301xxxxx at fromvoipfone:1] Set("SIP/301xxxxx-09f74a00", "oh=0") in new
> stack
> [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
> [301xxxxx at fromvoipfone:2] NoOp("SIP/301xxxxx-09f74a00", "01295259352")
> in new stack
> [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
> [301xxxxx at fromvoipfone:3] GotoIf("SIP/301xxxxx-09f74a00",
> "0?bankhols|200|1") in new stack
> [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
> [301xxxxx at 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
> [100 at 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
> [100 at day:2] Set("SIP/301xxxxx-09f74a00", "CALLERID(name)=xxxx") in new
> stack
> [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
> [100 at day:3] Macro("SIP/301xxxxx-09f74a00", "monitor|01327xxxxxx|"in"")
> in new stack
> [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
> [s at 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
> [s at 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
> [100 at 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
>