Re: [asterisk-users] caller getting cut off intermittently

2010-01-19 Thread John Taylor
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=xx
 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/301x-09f74a00, oh=0) in new
 stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [301xx...@fromvoipfone:2] NoOp(SIP/301x-09f74a00, 01295259352)
 in new stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [301xx...@fromvoipfone:3] GotoIf(SIP/301x-09f74a00,
 0?bankhols|200|1) in new stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [301xx...@fromvoipfone:4] GotoIfTime(SIP/301x-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/301x-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/301x-09f74a00, CALLERID(name)=) in new
 stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [...@day:3] Macro(SIP/301x-09f74a00, monitor|01327xx|in)
 in new stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [...@macro-monitor:1] Set(SIP/301x-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/301x-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/301x-09f74a00,
 SIP/203SIP/204SIP/206SIP/207SIP/220SIP/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/301x-09f74a00



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Re: [asterisk-users] caller getting cut off intermittently

2010-01-19 Thread John Taylor
Hi all,

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

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=xx
 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/301x-09f74a00, oh=0) in new
 stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [301xx...@fromvoipfone:2] NoOp(SIP/301x-09f74a00, 01295259352)
 in new stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [301xx...@fromvoipfone:3] GotoIf(SIP/301x-09f74a00,
 0?bankhols|200|1) in new stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [301xx...@fromvoipfone:4] GotoIfTime(SIP/301x-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/301x-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/301x-09f74a00, CALLERID(name)=) in new
 stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [...@day:3] Macro(SIP/301x-09f74a00, monitor|01327xx|in)
 in new stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [...@macro-monitor:1] Set(SIP/301x-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/301x-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/301x-09f74a00,
 SIP/203SIP/204SIP/206SIP/207SIP/220SIP/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/301x-09f74a00


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Re: [asterisk-users] caller getting cut off intermittently

2010-01-07 Thread 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 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=xx
 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/301x-09f74a00, oh=0) in new
 stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [301xx...@fromvoipfone:2] NoOp(SIP/301x-09f74a00, 01295259352)
 in new stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [301xx...@fromvoipfone:3] GotoIf(SIP/301x-09f74a00,
 0?bankhols|200|1) in new stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [301xx...@fromvoipfone:4] GotoIfTime(SIP/301x-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/301x-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/301x-09f74a00, CALLERID(name)=) in new
 stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [...@day:3] Macro(SIP/301x-09f74a00, monitor|01327xx|in)
 in new stack
 [Jan  4 09:58:56] VERBOSE[10712] logger.c:     -- Executing
 [...@macro-monitor:1] Set(SIP/301x-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/301x-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/301x-09f74a00,
 SIP/203SIP/204SIP/206SIP/207SIP/220SIP/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/301x-09f74a00


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