thr3ads.net - asterisk users - SOLVED: Re: [Asterisk-Users] Calling Extensions generates congestion when call answered [Jul 2006]

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Levis Kimotho
2006-Jul-06 06:25 UTC
SOLVED: Re: [Asterisk-Users] Calling Extensions generates congestion when call answered

Hi,

I installed G 729 and G 723 codecs and it works like magic. Download
link http://kvin.lv/pub/Linux/Asterisk/built-for-asterisk-1.2-untested/


-Kim

On 7/4/06, Levis Kimotho <kimy.voip@gmail.com>
wrote:>
> 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&127.0.0.1/255.255.255.0is 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@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@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@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@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@192.168.1.41' of Response 2:
> Match Found
> Jul 4 16:38:14 DEBUG[4873] chan_sip.c: Stopping retransmission on '
> 2d22a60a405995d45fef029904d63888@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@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@local.xorcom.com
> > icq#16849755       iax:tzafrir@local.xorcom.com
> > +972-50-7952406
> > tzafrir.cohen@xorcom.com   http://www.xorcom.com
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
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asterisk users - Jul 2006 - SOLVED: Re: Calling Extensions generates congestion when call answered

SOLVED: Re: [Asterisk-Users] Calling Extensions generates congestion when call answered
