OK, went thru the dialplan and it is identical to the 3.10 version. Ran the tail command and got the log info:
"2009-05-05T18:24:00.355374Z":1141:OUTGOING:INFO:sipex.redphonetech.local:SipClientTcp-18:B6570B90:SipXProxy:"SipUserAgent::sendUdp UDP SIP User Agent sent message:\n----Remote Host:192.168.254.102---- Port: 5060----\nSIP/2.0 404 Not Found\r\nFrom: \"Help Line\" <sip:4...@192.168.254.102>;tag=as081d8263\r\nTo: <sip:2...@192.168.254.120>;tag=e9caf392\r\nCall-Id: 038f7246200589c9748165814549a...@192.168.254.102\r\ncseq: 102 INVITE\r\nVia: SIP/2.0/UDP 192.168.254.102:5060;branch=z9hG4bK026ae390;rport=5060\r\nRecord-Route: <sip:192.168.254.120:5060;lr>\r\nUser-Agent: sipXecs/4.0.0 sipXecs/registry (Linux)\r\nDate: Tue, 05 May 2009 18:24:00 GMT\r\nAllow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, REGISTER, SUBSCRIBE\r\nAccept-Language: en\r\nSupported: gruu, path\r\nContent-Length: 0\r\n\r\n--------------------END--------------------" "2009-05-05T18:24:00.355897Z":1142:INCOMING:INFO:sipex.redphonetech.local:SipClientUdp-8:B78FCB90:SipXProxy:"Read SIP message:\n----Remote Host:192.168.254.102---- Port: 5060----\nACK sip:2...@192.168.254.120 SIP/2.0\r\nVia: SIP/2.0/UDP 192.168.254.102:5060;branch=z9hG4bK026ae390;rport\r\nFrom: \"Help Line\" <sip:4...@192.168.254.102>;tag=as081d8263\r\nTo: <sip:2...@192.168.254.120>;tag=e9caf392\r\nContact: <sip:4...@192.168.254.102>\r\nCall-ID: 038f7246200589c9748165814549a...@192.168.254.102\r\ncseq: 102 ACK\r\nUser-Agent: Asterisk PBX\r\nMax-Forwards: 70\r\nContent-Length: 0\r\n\r\n====================END====================" "2009-05-05T18:24:00.356434Z":1143:SIP:DEBUG:sipex.redphonetech.local:SipClientUdp-8:B78FCB90:SipXProxy:"SipTransactionList::findTransactionFor ACK match for message 0x8683518((nil)) INCOMING current match (0xb5f00950) ACK previous match ((nil)) UNKNOWN" "2009-05-05T18:24:00.356509Z":1144:SIP:DEBUG:sipex.redphonetech.local:SipClientUdp-8:B78FCB90:SipXProxy:"SipUserAgent[SipUserAgent-2]::dispatch transaction = 0xb5f00950, relationship = 10" "2009-05-05T18:24:00.356546Z":1145:SIP:DEBUG:sipex.redphonetech.local:SipClientUdp-8:B78FCB90:SipXProxy:"SipTransaction::handleIncoming 0xb5f00950 relationship ACK" "2009-05-05T18:24:00.356600Z":1146:SIP:DEBUG:sipex.redphonetech.local:SipClientUdp-8:B78FCB90:SipXProxy:"SipTransaction::handleChildIncoming 0xb5f00950 relationship ACK parent (nil)" "2009-05-05T18:24:00.368771Z":1147:ODBC:DEBUG:sipex.redphonetech.local:SipXProxyCseObserver-10:B76FAB90:SipXProxy:"odbcExecute: statement INSERT INTO call_state_events VALUES (DEFAULT,'192.168.254.120:-1',5,timestamp '2009-05-05 18:24:00.345','R',102,'038f7246200589c9748165814549a...@192.168.254.102','as081d8263','','\"Help Line\" <sip:4...@192.168.254.102>;tag=as081d8263','<sip:2...@192.168.254.120>','','','',0,'',''); succeeded" "2009-05-05T18:24:00.368806Z":1148:CDR:DEBUG:sipex.redphonetech.local:SipXProxyCseObserver-10:B76FAB90:SipXProxy:"CallStateEventWriter_DB::writeLog" "2009-05-05T18:24:00.368965Z":1149:SIP:DEBUG:sipex.redphonetech.local:SipXProxyCseObserver-10:B76FAB90:SipXProxy:"SipXProxyCseObserver message is a Call Failure" "2009-05-05T18:24:00.374678Z":1150:ODBC:DEBUG:sipex.redphonetech.local:SipXProxyCseObserver-10:B76FAB90:SipXProxy:"odbcExecute: statement INSERT INTO call_state_events VALUES (DEFAULT,'192.168.254.120:-1',6,timestamp '2009-05-05 18:24:00.368','F',102,'038f7246200589c9748165814549a...@192.168.254.102','as081d8263','e9caf392','\"Help Line\" <sip:4...@192.168.254.102>;tag=as081d8263','<sip:2...@192.168.254.120>;tag=e9caf392','','','',404,'Not Found',''); succeeded" "2009-05-05T18:24:00.374716Z":1151:CDR:DEBUG:sipex.redphonetech.local:SipXProxyCseObserver-10:B76FAB90:SipXProxy:"CallStateEventWriter_DB::writeLog" "2009-05-05T18:24:00.451039Z":1152:SIP:DEBUG:sipex.redphonetech.local:SipUserAgent-2:B6C77B90:SipXProxy:"SipUserAgent[SipUserAgent-2]::handleMessage resend Timeout of message for 0 protocol, callId: \"038f7246200589c9748165814549a...@192.168.254.102\"" "2009-05-05T18:24:00.451174Z":1153:SIP:DEBUG:sipex.redphonetech.local:SipUserAgent-2:B6C77B90:SipXProxy:"SipTransaction::handleResendEvent no response, TRANSACTION_COMPLETE" "2009-05-05T18:24:00.451297Z":1154:SIP:DEBUG:sipex.redphonetech.local:SipUserAgent-2:B6C77B90:SipXProxy:"SipUserAgent[SipUserAgent-2]::handleMessage calling garbageCollection()" "2009-05-05T18:24:00.457017Z":1155:SIP:DEBUG:sipex.redphonetech.local:SipUserAgent-2:B6C77B90:SipXProxy:"SipUserAgent[SipUserAgent-2]::handleMessage resend Timeout of message for 1 protocol, callId: \"038f7246200589c9748165814549a...@192.168.254.102\"" "2009-05-05T18:24:00.457289Z":1156:SIP:DEBUG:sipex.redphonetech.local:SipUserAgent-2:B6C77B90:SipXProxy:"SipUserAgent[SipUserAgent-2]::handleMessage calling garbageCollection()" "2009-05-05T18:24:01.551877Z":1157:SIP:DEBUG:sipex.redphonetech.local:SipUserAgent-2:B6C77B90:SipXProxy:"SipUserAgent[SipUserAgent-2]::handleMessage calling garbageCollection()" "2009-05-05T18:24:01.644873Z":1158:SIP:DEBUG:sipex.redphonetech.local:SipUserAgent-2:B6C77B90:SipXProxy:"SipUserAgent[SipUserAgent-2]::handleMessage transaction expiration message received" "2009-05-05T18:24:01.645099Z":1159:SIP:DEBUG:sipex.redphonetech.local:SipUserAgent-2:B6C77B90:SipXProxy:"SipUserAgent[SipUserAgent-2]::handleMessage calling garbageCollection()" The only thing that I see is the "404 not found" -----Original Message----- From: Tony Graziano [mailto:tgrazi...@myitdepartment.net] Sent: Tuesday, May 05, 2009 10:41 AM To: sipx-users@list.sipfoundry.org; Mark Wood Subject: Re: [sipx-users] Unmanged Gateways I would suggest you go through your dialplan entries and compare them to what you had before. You might also think of putting a "catchall" rule in for troubleshooting and turn the proxy log up to debug and see what is being sent to sipx in the way of digits. I saw a "weird" thing too trying to return calls through our Ingate, which is also an unmanaged gateway. I used to have an entry "%2B" that would have to be stripped on returning calls from call-logs on the phone where they showed "+" (and an eleven digit number). In 4.0 that call needs to now send "+" instead of the SIP HEX value of of + as "%2B" in order for it to make it past the dial rule and hit the proxy and get sent. At that point if you tail the log: tail -f /var/log/sipxpbx/sipXproxy.log you might be able to see what the proxy might be complaining about. So In my case I changed the rule from a prefix of %2B to + and now it all works again. Dialing 1 (and 11 digits) always worked, but when the callerid said +1 and you wanted to hit the dial button on the polycom it did not until I made the above change. So I am convinced some items did change in the proxy for sure. >>> Mark Wood <mark.w...@redphonetech.com> 05/05/09 12:54 PM >>> Fresh ISO install of 4.0 Setup an Asterisk box as an unmanaged gateway for interim purposes, also setup associated dial plan. I can make calls from the SipXecs system to asterisk and use the Asterisk box as a PSTN gateway as well. But calls from the Asterisk to the SipXecs do not go through, it (Asterisk) responds with an "all circuits are busy" announcement. If I check the CDR records in the SipXecs system it shows the call with all the correct info but with a failed in the status column. This exact configuration worked in our 3.10.2 system, any additional configs for the 4,0? M Wood _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
