[Asterisk-Users] Calling Extensions generates congestion when
call answered
Levis Kimotho
kimy.voip at gmail.com
Tue Jul 4 06:43:33 MST 2006
Hi,
Below is part of the log file
Jul 4 16:38:06 VERBOSE[5871] logger.c: dialparties.agi: Caller ID name is
'LAN201' number is '1235'
Jul 4 16:38:06 VERBOSE[5871] logger.c: dialparties.agi: Methodology of ring
is 'none'
Jul 4 16:38:06 VERBOSE[5871] logger.c: -- dialparties.agi: Added extension
8888 to extension map
Jul 4 16:38:06 DEBUG[5871] db.c: Unable to find key '8888' in family 'CF'
Jul 4 16:38:06 VERBOSE[5871] logger.c: -- dialparties.agi: Extension 8888 cf
is disabled
Jul 4 16:38:06 DEBUG[5871] db.c: Unable to find key '8888' in family 'DND'
Jul 4 16:38:06 VERBOSE[5871] logger.c: -- dialparties.agi: Extension 8888 do
not disturb is disabled
Jul 4 16:38:06 DEBUG[5871] db.c: Unable to find key '8888' in family 'CW'
Jul 4 16:38:06 DEBUG[5871] db.c: Unable to find key '8888' in family 'CFB'
Jul 4 16:38:06 DEBUG[5871] db.c: Unable to find key '8888' in family 'CFU'
Jul 4 16:38:06 DEBUG[5876] manager.c: Manager received command 'login'
Jul 4 16:38:06 VERBOSE[5876] logger.c: == Parsing
'/etc/asterisk/manager.conf': Jul 4 16:38:06 VERBOSE[5876] logger.c: ==
Parsing '/etc/asterisk/manager.conf': Found
Jul 4 16:38:06 VERBOSE[5876] logger.c: == Parsing
'/etc/asterisk/manager_additional.conf': Jul 4 16:38:06 VERBOSE[5876]
logger.c: == Parsing '/etc/asterisk/manager_additional.conf': Found
Jul 4 16:38:06 DEBUG[5876] acl.c: 0.0.0.0/0.0.0.0/0.0.0.0 appended to acl
for peer
Jul 4 16:38:06 WARNING[5876] acl.c: 255.255.255.0&127.0.0.1/255.255.255.0 is
not a valid netmask
Jul 4 16:38:06 VERBOSE[5876] logger.c: == Manager 'admin' logged on from
127.0.0.1
Jul 4 16:38:06 DEBUG[5876] manager.c: Manager received command
'ExtensionState'
Jul 4 16:38:06 DEBUG[5876] manager.c: Manager received command 'Logoff'
Jul 4 16:38:06 VERBOSE[5871] logger.c: -- dialparties.agi: Checking CW and
CFB status for extension 8888
Jul 4 16:38:06 VERBOSE[5876] logger.c: == Manager 'admin' logged off from
127.0.0.1
Jul 4 16:38:06 VERBOSE[5871] logger.c: -- dialparties.agi: DbSet
CALLTRACE/8888 to 1235
Jul 4 16:38:06 VERBOSE[5871] logger.c: -- AGI Script
dialparties.agicompleted, returning 0
Jul 4 16:38:06 VERBOSE[5871] logger.c: -- Executing Dial("SIP/1235-220e",
"SIP/8888|15|tr") in new stack
Jul 4 16:38:06 DEBUG[5871] chan_sip.c: Setting NAT on RTP to 0
Jul 4 16:38:06 DEBUG[5871] chan_sip.c: Outgoing Call for 8888
Jul 4 16:38:06 VERBOSE[5871] logger.c: -- Called 8888
Jul 4 16:38:06 DEBUG[4873] chan_sip.c: (Provisional) Stopping retransmission
(but retaining packet) on '491b3f4b5b6d3d58764c458055e01906 at 192.168.1.41'
Request 102: Found
Jul 4 16:38:06 DEBUG[4873] chan_sip.c: (Provisional) Stopping retransmission
(but retaining packet) on '491b3f4b5b6d3d58764c458055e01906 at 192.168.1.41'
Request 102: Found
Jul 4 16:38:06 VERBOSE[5871] logger.c: -- SIP/8888-bde7 is ringing
Jul 4 16:38:10 DEBUG[4873] chan_sip.c: Acked pending invite 102
Jul 4 16:38:10 DEBUG[4873] chan_sip.c: Stopping retransmission on '
491b3f4b5b6d3d58764c458055e01906 at 192.168.1.41' of Request 102: Match Found
Jul 4 16:38:10 DEBUG[4873] chan_sip.c: Oooh, we need to change our formats
since our peer supports only 0x1 (g723) and not 0x4 (ulaw)
Jul 4 16:38:10 WARNING[4873] channel.c: Unable to find a codec translation
path from g723 to ulaw
Jul 4 16:38:10 WARNING[4873] channel.c: Unable to find a codec translation
path from g723 to ulaw
Jul 4 16:38:10 DEBUG[4873] chan_sip.c: build_route: Contact hop: Muriuki
Jul 4 16:38:10 VERBOSE[5871] logger.c: -- SIP/8888-bde7 answered
SIP/1235-220e
Jul 4 16:38:10 WARNING[5871] channel.c: No path to translate from
SIP/1235-220e(4) to SIP/8888-bde7(1)
Jul 4 16:38:10 WARNING[5871] app_dial.c: Had to drop call because I couldn't
make SIP/1235-220e compatible with SIP/8888-bde7
Jul 4 16:38:10 DEBUG[5871] chan_sip.c: update_call_counter(8888) - decrement
call limit counter
Jul 4 16:38:10 VERBOSE[5871] logger.c: == Spawn extension (macro-dial, s,
10) exited non-zero on 'SIP/1235-220e' in macro 'dial'
Jul 4 16:38:10 DEBUG[4873] chan_sip.c: Stopping retransmission on '
491b3f4b5b6d3d58764c458055e01906 at 192.168.1.41' of Request 103: Match Found
Jul 4 16:38:10 VERBOSE[5871] logger.c: == Spawn extension (macro-dial, s,
10) exited non-zero on 'SIP/1235-220e' in macro 'exten-vm'
Jul 4 16:38:10 VERBOSE[5871] logger.c: == Spawn extension (macro-dial, s,
10) exited non-zero on 'SIP/1235-220e'
Jul 4 16:38:10 DEBUG[5871] cdr_addon_mysql.c: cdr_mysql: inserting a CDR
record.
Jul 4 16:38:10 DEBUG[5871] cdr_addon_mysql.c: cdr_mysql: SQL command as
follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid)
VALUES ('2006-07-04 16:38:02','\"LAN201\"
<1235>','1235','8888','from-internal',
'SIP/1235-220e','SIP/8888-bde7','Dial','SIP/8888|15|tr',8,0,'NO
ANSWER',3,'1235','1152020282.0')
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is '"LAN201" <1235>'
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is '1235'
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is '8888'
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is 'from-internal'
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is 'SIP/1235-220e'
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is 'SIP/8888-bde7'
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is 'Dial'
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is 'SIP/8888|15|tr'
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is '2006-07-04 16:38:02'
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is '(null)'
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is '2006-07-04 16:38:10'
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is '8'
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is '0'
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is 'NO ANSWER'
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is 'DOCUMENTATION'
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is '1235'
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is '1152020282.0'
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is '(null)'
Jul 4 16:38:10 DEBUG[5871] chan_sip.c: update_call_counter(1235) - decrement
call limit counter
Jul 4 16:38:10 DEBUG[5871] chan_sip.c: AST hangup cause 16 (no match found
in SIP)
Jul 4 16:38:10 DEBUG[4873] chan_sip.c: Stopping retransmission on '
515e6c-c0a8018c-13c4-44aa9991-5394022-303b at 192.168.1.41' of Response 2:
Match Found
Jul 4 16:38:14 DEBUG[4873] chan_sip.c: Stopping retransmission on '
2d22a60a405995d45fef029904d63888 at 192.168.1.41' of Request 102: Match Found
Jul 4 16:38:26 DEBUG[4873] chan_sip.c: Auto destroying call
'c0a8018c-13c4-0-2391-4282'
Jul 4 16:38:26 DEBUG[4873] chan_sip.c: Auto destroying call
'c0a8018c-13c4-0-2391-4282'
-Kim
On 7/4/06, Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote:
>
> On Tue, Jul 04, 2006 at 01:49:31PM +0300, Levis Kimotho wrote:
> > Hi,
> >
> > I just installed freePBX n Asterisk (Fedora 5, ast*1.2.9.1)and they are
> > working well except when i created 2 extensions i.e 8888 n 1235, when i
> try
> > to call either from my SIP Phones, when i pick the call from one of the
> > extension, the call fails and i hear a ¨busy tone¨. Another problem
> > arrises
> > when if the call dials for more than 10s, the call fails and generates a
> > busy tone. Ive attached my log file
>
> No, you haven't. Or maybe it was cut away by the list server.
>
> In that case, add a small call trace inline.
>
> --
> Tzafrir Cohen sip:tzafrir at local.xorcom.com
> icq#16849755 iax:tzafrir at local.xorcom.com
> +972-50-7952406
> tzafrir.cohen at xorcom.com http://www.xorcom.com
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