Hi,

I've now set dtmfmode=rfc2833 and that seems to have fixed it

John


2010/1/7 John Taylor <j...@vetsurgeon.org.uk>:
> 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
>>
>

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