Hi,<br><br>Below is part of the log file<br><br>
Jul 4 16:38:06 VERBOSE[5871] logger.c: dialparties.agi: Caller ID name is 'LAN201' number is '1235'<br>
Jul 4 16:38:06 VERBOSE[5871] logger.c: dialparties.agi: Methodology of ring is 'none'<br>
Jul 4 16:38:06 VERBOSE[5871] logger.c: -- dialparties.agi: Added extension 8888 to extension map<br>
Jul 4 16:38:06 DEBUG[5871] db.c: Unable to find key '8888' in family 'CF'<br>
Jul 4 16:38:06 VERBOSE[5871] logger.c: -- dialparties.agi: Extension 8888 cf is disabled<br>
Jul 4 16:38:06 DEBUG[5871] db.c: Unable to find key '8888' in family 'DND'<br>
Jul 4 16:38:06 VERBOSE[5871] logger.c: -- dialparties.agi: Extension 8888 do not disturb is disabled<br>
Jul 4 16:38:06 DEBUG[5871] db.c: Unable to find key '8888' in family 'CW'<br>
Jul 4 16:38:06 DEBUG[5871] db.c: Unable to find key '8888' in family 'CFB'<br>
Jul 4 16:38:06 DEBUG[5871] db.c: Unable to find key '8888' in family 'CFU'<br>
Jul 4 16:38:06 DEBUG[5876] manager.c: Manager received command 'login'<br>
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<br>
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<br>
Jul 4 16:38:06 DEBUG[5876] acl.c: <a href="http://0.0.0.0/0.0.0.0/0.0.0.0">0.0.0.0/0.0.0.0/0.0.0.0</a> appended to acl for peer<br>
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<br>
Jul 4 16:38:06 VERBOSE[5876] logger.c: == Manager 'admin' logged on from <a href="http://127.0.0.1">127.0.0.1</a><br>
Jul 4 16:38:06 DEBUG[5876] manager.c: Manager received command 'ExtensionState'<br>
Jul 4 16:38:06 DEBUG[5876] manager.c: Manager received command 'Logoff'<br>
Jul 4 16:38:06 VERBOSE[5871] logger.c: -- dialparties.agi: Checking CW and CFB status for extension 8888<br>
Jul 4 16:38:06 VERBOSE[5876] logger.c: == Manager 'admin' logged off from <a href="http://127.0.0.1">127.0.0.1</a><br>
Jul 4 16:38:06 VERBOSE[5871] logger.c: -- dialparties.agi: DbSet CALLTRACE/8888 to 1235<br>
Jul 4 16:38:06 VERBOSE[5871] logger.c: -- AGI Script dialparties.agi completed, returning 0<br>
Jul 4 16:38:06 VERBOSE[5871] logger.c: -- Executing Dial("SIP/1235-220e", "SIP/8888|15|tr") in new stack<br>
Jul 4 16:38:06 DEBUG[5871] chan_sip.c: Setting NAT on RTP to 0<br>
Jul 4 16:38:06 DEBUG[5871] chan_sip.c: Outgoing Call for 8888<br>
Jul 4 16:38:06 VERBOSE[5871] logger.c: -- Called 8888<br>
Jul 4 16:38:06 DEBUG[4873] chan_sip.c: (Provisional) Stopping
retransmission (but retaining packet) on
'<a href="mailto:491b3f4b5b6d3d58764c458055e01906@192.168.1.41">491b3f4b5b6d3d58764c458055e01906@192.168.1.41</a>' Request 102: Found<br>
Jul 4 16:38:06 DEBUG[4873] chan_sip.c: (Provisional) Stopping
retransmission (but retaining packet) on
'<a href="mailto:491b3f4b5b6d3d58764c458055e01906@192.168.1.41">491b3f4b5b6d3d58764c458055e01906@192.168.1.41</a>' Request 102: Found<br>
Jul 4 16:38:06 VERBOSE[5871] logger.c: -- SIP/8888-bde7 is ringing<br>
Jul 4 16:38:10 DEBUG[4873] chan_sip.c: Acked pending invite 102<br>
Jul 4 16:38:10 DEBUG[4873] chan_sip.c: Stopping retransmission on
'<a href="mailto:491b3f4b5b6d3d58764c458055e01906@192.168.1.41">491b3f4b5b6d3d58764c458055e01906@192.168.1.41</a>' of Request 102: Match
Found<br>
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)<br>
Jul 4 16:38:10 WARNING[4873] channel.c: Unable to find a codec translation path from g723 to ulaw<br>
Jul 4 16:38:10 WARNING[4873] channel.c: Unable to find a codec translation path from g723 to ulaw<br>
Jul 4 16:38:10 DEBUG[4873] chan_sip.c: build_route: Contact hop: Muriuki <br>
Jul 4 16:38:10 VERBOSE[5871] logger.c: -- SIP/8888-bde7 answered SIP/1235-220e<br>
Jul 4 16:38:10 WARNING[5871] channel.c: No path to translate from SIP/1235-220e(4) to SIP/8888-bde7(1)<br>
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<br>
Jul 4 16:38:10 DEBUG[5871] chan_sip.c: update_call_counter(8888) - decrement call limit counter<br>
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'<br>
Jul 4 16:38:10 DEBUG[4873] chan_sip.c: Stopping retransmission on
'<a href="mailto:491b3f4b5b6d3d58764c458055e01906@192.168.1.41">491b3f4b5b6d3d58764c458055e01906@192.168.1.41</a>' of Request 103: Match
Found<br>
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'<br>
Jul 4 16:38:10 VERBOSE[5871] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/1235-220e'<br>
Jul 4 16:38:10 DEBUG[5871] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record.<br>
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')<br>
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is '"LAN201" <1235>'<br>
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is '1235'<br>
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is '8888'<br>
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is 'from-internal'<br>
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is 'SIP/1235-220e'<br>
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is 'SIP/8888-bde7'<br>
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is 'Dial'<br>
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is 'SIP/8888|15|tr'<br>
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is '2006-07-04 16:38:02'<br>
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is '(null)'<br>
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is '2006-07-04 16:38:10'<br>
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is '8'<br>
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is '0'<br>
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is 'NO ANSWER'<br>
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is 'DOCUMENTATION'<br>
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is '1235'<br>
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is '1152020282.0'<br>
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is '(null)'<br>
Jul 4 16:38:10 DEBUG[5871] chan_sip.c: update_call_counter(1235) - decrement call limit counter<br>
Jul 4 16:38:10 DEBUG[5871] chan_sip.c: AST hangup cause 16 (no match found in SIP)<br>
Jul 4 16:38:10 DEBUG[4873] chan_sip.c: Stopping retransmission on
'<a href="mailto:515e6c-c0a8018c-13c4-44aa9991-5394022-303b@192.168.1.41">515e6c-c0a8018c-13c4-44aa9991-5394022-303b@192.168.1.41</a>' of Response
2: Match Found<br>
Jul 4 16:38:14 DEBUG[4873] chan_sip.c: Stopping retransmission on
'<a href="mailto:2d22a60a405995d45fef029904d63888@192.168.1.41">2d22a60a405995d45fef029904d63888@192.168.1.41</a>' of Request 102: Match
Found<br>
Jul 4 16:38:26 DEBUG[4873] chan_sip.c: Auto destroying call 'c0a8018c-13c4-0-2391-4282'<br>
Jul 4 16:38:26 DEBUG[4873] chan_sip.c: Auto destroying call 'c0a8018c-13c4-0-2391-4282'<br><br>-Kim<br><br><div><span class="gmail_quote">On 7/4/06, <b class="gmail_sendername">Tzafrir Cohen</b> <<a href="mailto:tzafrir.cohen@xorcom.com">
tzafrir.cohen@xorcom.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">On Tue, Jul 04, 2006 at 01:49:31PM +0300, Levis Kimotho wrote:
<br>> Hi,<br>><br>> I just installed freePBX n Asterisk (Fedora 5, ast*<a href="http://1.2.9.1">1.2.9.1</a>)and they are<br>> working well except when i created 2 extensions i.e 8888 n 1235, when i try<br>> to call either from my SIP Phones, when i pick the call from one of the
<br>> extension, the call fails and i hear a ¨busy tone¨. Another problem<br>> arrises<br>> when if the call dials for more than 10s, the call fails and generates a<br>> busy tone. Ive attached my log file<br>
<br>No, you haven't. Or maybe it was cut away by the list server.<br><br>In that case, add a small call trace inline.<br><br>--<br>Tzafrir Cohen <a href="mailto:sip:tzafrir@local.xorcom.com">sip:tzafrir@local.xorcom.com
</a><br>icq#16849755 <a href="mailto:iax:tzafrir@local.xorcom.com">iax:tzafrir@local.xorcom.com</a><br>+972-50-7952406<br><a href="mailto:tzafrir.cohen@xorcom.com">tzafrir.cohen@xorcom.com</a> <a href="http://www.xorcom.com">
http://www.xorcom.com</a><br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br><br>Asterisk-Users mailing list<br>To UNSUBSCRIBE or update options visit:
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