http://www.mhspot.com/stsforum/index.php Fri, 03 Sep 2010 10:18:17 +0000 PunBB http://www.mhspot.com/stsforum/viewtopic.php?id=345&action=new From: "73003" ;tag=9c6a0606 Call-ID: MjI5YzhkZTQ1OGVmYzA1Y2NkMjBmODVmNTE1ZjA5Y2M. CSeq: 1 INVITE Server: mjsip stack 1.6 sts.2 Content-Length: 0 Our SIP A/S complains at the To header as it doesnt contain a tag to identify the callee end point. After looking at the SIP Spec we have deduced that the response To header should have a tag, and as such, this would seem to be a bug with the gateway. Additionally, is 603 the best response to send. What if we fork out to more than one sykpe client, where multiple clients could potentially answer the initial INVITE. A 603 would not be valid in such a case. Would a 486 be a better response???

]]> Fri, 03 Sep 2010 10:18:17 +0000 http://www.mhspot.com/stsforum/viewtopic.php?id=345&action=new http://www.mhspot.com/stsforum/viewtopic.php?id=344&action=new Mon, 30 Aug 2010 14:20:50 +0000 http://www.mhspot.com/stsforum/viewtopic.php?id=344&action=new http://www.mhspot.com/stsforum/viewtopic.php?id=305&action=new /dev/null & #Start stsTrunk cd /usr/share/siptosis nohup ./stsTrunk_Control start >/dev/null & ;; stop) #Stop stsProxy cd /usr/share/stsproxy nohup ./stsProxy_Control stop >/dev/null & #Stop stsTrunk cd /usr/share/siptosis nohup ./stsTrunk_Control stop >/dev/null & ;; restart) $0 stop $0 start ;; status) #Status stsProxy cd /usr/share/stsproxy ./stsProxy_Control status #Status stsTrunk cd /usr/share/siptosis ./stsTrunk_Control status ;; *) echo "Usage: $0 {start|stop|restart|status}" exit 1 ;; esac exit 0 [/code]]]> Fri, 27 Aug 2010 01:48:59 +0000 http://www.mhspot.com/stsforum/viewtopic.php?id=305&action=new http://www.mhspot.com/stsforum/viewtopic.php?id=343&action=new Thu, 26 Aug 2010 12:54:02 +0000 http://www.mhspot.com/stsforum/viewtopic.php?id=343&action=new http://www.mhspot.com/stsforum/viewtopic.php?id=342&action=new
]]> Mon, 23 Aug 2010 13:32:39 +0000 http://www.mhspot.com/stsforum/viewtopic.php?id=342&action=new http://www.mhspot.com/stsforum/viewtopic.php?id=341&action=new Sat, 21 Aug 2010 16:01:51 +0000 http://www.mhspot.com/stsforum/viewtopic.php?id=341&action=new http://www.mhspot.com/stsforum/viewtopic.php?id=340&action=new Fri, 20 Aug 2010 12:42:57 +0000 http://www.mhspot.com/stsforum/viewtopic.php?id=340&action=new http://www.mhspot.com/stsforum/viewtopic.php?id=339&action=new 2010-08-16 22:35:19,481 RTP Ports: 63200-63200 Local Skype Ports: 64432-64433 2010-08-16 22:35:19,631 WAITING FOR INCOMING CALL 2010-08-17 10:34:38,677 callReceived - incoming Skype Call from:Jane Doe [jane.doe] status:RINGING 2010-08-17 10:34:38,683 callReceived - Direct Sip Dial to:sip:75973@127.0.0.1:5060 from:"jane.doe" 2010-08-17 10:34:38,683 UAC: CALLING sip:75973@127.0.0.1:5060 2010-08-17 10:34:38,701 Attempting call From:"jane.doe" To: 2010-08-17 10:34:38,706 skypeCallStatus - RINGING 2010-08-17 10:34:41,500 ACCEPTED/CALL 2010-08-17 10:34:43,231 Starting codec: PCMU/8000(0) rfc2833Type(101) 2010-08-17 10:34:43,282 updating RTP media destination to:127.0.0.1:13604 from:75.xx.xx.164:13604 <==PUBLIC IP OUTSIDE NAT 2010-08-17 10:34:43,293 RtpSender Address locked. 2010-08-17 10:34:43,327 skypeCallStatus - INPROGRESS 2010-08-17 10:35:35,040 rtp Timeout: hanging up 2010-08-17 10:35:35,045 SkypeAudioSender stats - packets:434 underRuns:266 maxLatency:20ms avgLatency:20ms 2010-08-17 10:35:35,051 RTPSender stats - packets:1868 underRuns:29 maxLatency:180ms avgLatency:104ms 2010-08-17 10:35:35,067 SkypeAudioReceiver stats - packets:696 2010-08-17 10:35:35,286 RTPReceiver stats - packets:434 timeOuts:272 2010-08-17 10:36:27,011 ToSIP: sip:75973@127.0.0.1:5060 FromSkype: jane.doe CallTime: 1:43 2010-08-17 10:36:27,013 WAITING FOR INCOMING CALL 2010-08-17 10:36:27,298 skypeCallStatus - Complete: FINISHED [/code] asterisk full [code] [2010-08-17 10:34:38] VERBOSE[2851] netsock.c: == Using SIP RTP TOS bits 184 [2010-08-17 10:34:38] VERBOSE[2851] netsock.c: == Using SIP RTP CoS mark 5 [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [75973@from-trunk:1] Set("SIP/skype-00000003", "__FROM_DID=75973") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [75973@from-trunk:2] ExecIf("SIP/skype-00000003", "0 ?Set(CALLERID(name)=Skype_Caller)") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [75973@from-trunk:3] Set("SIP/skype-00000003", "__CALLINGPRES_SV=allowed_not_screened") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [75973@from-trunk:4] Set("SIP/skype-00000003", "CALLERPRES()=allowed_not_screened") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [75973@from-trunk:5] Goto("SIP/skype-00000003", "ext-group,700,1") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Goto (ext-group,700,1) [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [700@ext-group:1] Macro("SIP/skype-00000003", "user-callerid,") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [s@macro-user-callerid:1] Set("SIP/skype-00000003", "AMPUSER=Skype_Caller") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [s@macro-user-callerid:2] GotoIf("SIP/skype-00000003", "0?report") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [s@macro-user-callerid:3] ExecIf("SIP/skype-00000003", "1?Set(REALCALLERIDNUM=Skype_Caller)") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [s@macro-user-callerid:4] Set("SIP/skype-00000003", "AMPUSER=") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [s@macro-user-callerid:5] Set("SIP/skype-00000003", "AMPUSERCIDNAME=") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [s@macro-user-callerid:6] GotoIf("SIP/skype-00000003", "1?report") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Goto (macro-user-callerid,s,9) [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [s@macro-user-callerid:9] GotoIf("SIP/skype-00000003", "0?continue") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [s@macro-user-callerid:10] Set("SIP/skype-00000003", "__TTL=64") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [s@macro-user-callerid:11] GotoIf("SIP/skype-00000003", "1?continue") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Goto (macro-user-callerid,s,18) [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [s@macro-user-callerid:18] NoOp("SIP/skype-00000003", "Using CallerID "jane.doe" ") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [700@ext-group:2] GotoIf("SIP/skype-00000003", "1?skipdb") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Goto (ext-group,700,4) [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [700@ext-group:4] Set("SIP/skype-00000003", "__NODEST=") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [700@ext-group:5] Set("SIP/skype-00000003", "__BLKVM_OVERRIDE=BLKVM/700/SIP/skype-00000003") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [700@ext-group:6] Set("SIP/skype-00000003", "__BLKVM_BASE=700") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [700@ext-group:7] Set("SIP/skype-00000003", "DB(BLKVM/700/SIP/skype-00000003)=TRUE") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [700@ext-group:8] Set("SIP/skype-00000003", "RRNODEST=") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [700@ext-group:9] Set("SIP/skype-00000003", "__NODEST=700") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [700@ext-group:10] GosubIf("SIP/skype-00000003", "0?sub-rgsetcid,s,1") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [700@ext-group:11] Set("SIP/skype-00000003", "RecordMethod=Group") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [700@ext-group:12] Macro("SIP/skype-00000003", "record-enable,701-702,Group") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [s@macro-record-enable:1] GotoIf("SIP/skype-00000003", "1?check") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Goto (macro-record-enable,s,4) [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [s@macro-record-enable:4] ExecIf("SIP/skype-00000003", "0?MacroExit()") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [s@macro-record-enable:5] GotoIf("SIP/skype-00000003", "1?Group:OUT") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Goto (macro-record-enable,s,6) [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [s@macro-record-enable:6] Set("SIP/skype-00000003", "LOOPCNT=2") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [s@macro-record-enable:7] Set("SIP/skype-00000003", "ITER=1") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [s@macro-record-enable:8] GotoIf("SIP/skype-00000003", "1?continue") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Goto (macro-record-enable,s,13) [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [s@macro-record-enable:13] Set("SIP/skype-00000003", "ITER=2") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [s@macro-record-enable:14] GotoIf("SIP/skype-00000003", "1?begin") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Goto (macro-record-enable,s,8) [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [s@macro-record-enable:8] GotoIf("SIP/skype-00000003", "1?continue") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Goto (macro-record-enable,s,13) [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [s@macro-record-enable:13] Set("SIP/skype-00000003", "ITER=3") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [s@macro-record-enable:14] GotoIf("SIP/skype-00000003", "0?begin") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [s@macro-record-enable:15] GotoIf("SIP/skype-00000003", "0?IN") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [s@macro-record-enable:16] ExecIf("SIP/skype-00000003", "1?MacroExit()") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [700@ext-group:13] Set("SIP/skype-00000003", "RingGroupMethod=ringall") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [700@ext-group:14] Macro("SIP/skype-00000003", "dial,20,m(default)t,701-702") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [s@macro-dial:1] GotoIf("SIP/skype-00000003", "1?dial") in new stack [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Goto (macro-dial,s,3) [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [s@macro-dial:3] AGI("SIP/skype-00000003", "dialparties.agi") in new stack [2010-08-17 10:34:38] VERBOSE[6461] res_agi.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi [2010-08-17 10:34:38] VERBOSE[6461] res_agi.c: dialparties.agi: Starting New Dialparties.agi [2010-08-17 10:34:38] VERBOSE[6461] res_agi.c: dialparties.agi: Caller ID name is 'jane.doe' number is 'Skype_Caller' [2010-08-17 10:34:38] VERBOSE[6461] res_agi.c: > dialparties.agi: USE_CONFIRMATION: 'FALSE' [2010-08-17 10:34:38] VERBOSE[6461] res_agi.c: > dialparties.agi: RINGGROUP_INDEX: '' [2010-08-17 10:34:38] VERBOSE[6461] res_agi.c: dialparties.agi: Methodology of ring is 'ringall' [2010-08-17 10:34:38] VERBOSE[6461] res_agi.c: -- dialparties.agi: Added extension 701 to extension map [2010-08-17 10:34:38] VERBOSE[6461] res_agi.c: -- dialparties.agi: Added extension 702 to extension map [2010-08-17 10:34:38] VERBOSE[6461] res_agi.c: -- dialparties.agi: Extension 701 cf is disabled [2010-08-17 10:34:38] VERBOSE[6461] res_agi.c: -- dialparties.agi: Extension 702 cf is disabled [2010-08-17 10:34:38] VERBOSE[6461] res_agi.c: -- dialparties.agi: Extension 701 do not disturb is disabled [2010-08-17 10:34:38] VERBOSE[6461] res_agi.c: -- dialparties.agi: Extension 702 do not disturb is disabled [2010-08-17 10:34:38] VERBOSE[6461] res_agi.c: > dialparties.agi: extnum 701 has: cw: 1; hascfb: 0 [] hascfu: 0 [] [2010-08-17 10:34:38] VERBOSE[6461] res_agi.c: -- dialparties.agi: DbDel CALLTRACE/701 - Caller ID is not defined [2010-08-17 10:34:38] VERBOSE[6461] res_agi.c: > dialparties.agi: extnum 702 has: cw: 1; hascfb: 0 [] hascfu: 0 [] [2010-08-17 10:34:38] VERBOSE[6461] res_agi.c: -- dialparties.agi: DbDel CALLTRACE/702 - Caller ID is not defined [2010-08-17 10:34:38] VERBOSE[6461] res_agi.c: -- dialparties.agi: Filtered ARG3: 701-702 [2010-08-17 10:34:38] VERBOSE[6461] res_agi.c: > dialparties.agi: NODEST: 700 adding M(auto-blkvm) to dialopts: m(default)tM(auto-blkvm) [2010-08-17 10:34:38] VERBOSE[6461] res_agi.c: > dialparties.agi: NODEST: 700 blkvm enabled macro already in dialopts: m(default)tM(auto-blkvm) [2010-08-17 10:34:38] VERBOSE[6461] res_agi.c: -- AGI Script dialparties.agi completed, returning 0 [2010-08-17 10:34:38] VERBOSE[6461] pbx.c: -- Executing [s@macro-dial:7] Dial("SIP/skype-00000003", "SIP/701&SIP/702,20,m(default)tM(auto-blkvm)") in new stack [2010-08-17 10:34:38] VERBOSE[6461] netsock.c: == Using SIP RTP TOS bits 184 [2010-08-17 10:34:38] VERBOSE[6461] netsock.c: == Using SIP RTP CoS mark 5 [2010-08-17 10:34:38] VERBOSE[6461] app_dial.c: -- Called 701 [2010-08-17 10:34:38] VERBOSE[6461] netsock.c: == Using SIP RTP TOS bits 184 [2010-08-17 10:34:38] VERBOSE[6461] netsock.c: == Using SIP RTP CoS mark 5 [2010-08-17 10:34:38] VERBOSE[6461] app_dial.c: -- Called 702 [2010-08-17 10:34:38] VERBOSE[6461] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/skype-00000003 [2010-08-17 10:34:38] VERBOSE[6461] app_dial.c: -- SIP/701-00000004 is ringing [2010-08-17 10:34:39] VERBOSE[6461] app_dial.c: -- SIP/702-00000005 is ringing [2010-08-17 10:34:41] VERBOSE[6461] app_dial.c: -- SIP/702-00000005 answered SIP/skype-00000003 [2010-08-17 10:34:41] VERBOSE[6461] pbx.c: -- Executing [s@macro-auto-blkvm:1] Set("SIP/702-00000005", "__MACRO_RESULT=") in new stack [2010-08-17 10:34:41] VERBOSE[6461] pbx.c: -- Executing [s@macro-auto-blkvm:2] NoOp("SIP/702-00000005", "Deleting: BLKVM/700/SIP/skype-00000003 TRUE") in new stack [2010-08-17 10:34:41] VERBOSE[6461] res_musiconhold.c: -- Stopped music on hold on SIP/skype-00000003 [2010-08-17 10:34:51] WARNING[2851] chan_sip.c: Maximum retries exceeded on transmission 238138041187@127.0.0.1 for seqno 1 (Critical Response) -- See doc/sip-retransmit.txt. [2010-08-17 10:34:51] WARNING[2851] chan_sip.c: Hanging up call 238138041187@127.0.0.1 - no reply to our critical packet (see doc/sip-retransmit.txt). [2010-08-17 10:34:51] VERBOSE[6461] pbx.c: -- Executing [h@macro-dial:1] Macro("SIP/skype-00000003", "hangupcall") in new stack [2010-08-17 10:34:51] VERBOSE[6461] pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf("SIP/skype-00000003", "1?skiprg") in new stack [2010-08-17 10:34:51] VERBOSE[6461] pbx.c: -- Goto (macro-hangupcall,s,4) [2010-08-17 10:34:51] VERBOSE[6461] pbx.c: -- Executing [s@macro-hangupcall:4] GotoIf("SIP/skype-00000003", "0?skipblkvm") in new stack [2010-08-17 10:34:51] VERBOSE[6461] pbx.c: -- Executing [s@macro-hangupcall:5] NoOp("SIP/skype-00000003", "Cleaning Up Block VM Flag: BLKVM/700/SIP/skype-00000003") in new stack [2010-08-17 10:34:51] VERBOSE[6461] pbx.c: -- Executing [s@macro-hangupcall:6] NoOp("SIP/skype-00000003", "Deleting: BLKVM/700/SIP/skype-00000003 ") in new stack [2010-08-17 10:34:51] VERBOSE[6461] pbx.c: -- Executing [s@macro-hangupcall:7] GotoIf("SIP/skype-00000003", "1?theend") in new stack [2010-08-17 10:34:51] VERBOSE[6461] pbx.c: -- Goto (macro-hangupcall,s,9) [2010-08-17 10:34:51] VERBOSE[6461] pbx.c: -- Executing [s@macro-hangupcall:9] Hangup("SIP/skype-00000003", "") in new stack [2010-08-17 10:34:51] VERBOSE[6461] app_macro.c: == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/skype-00000003' in macro 'hangupcall' [2010-08-17 10:34:51] VERBOSE[6461] app_macro.c: == Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/skype-00000003' in macro 'dial' [2010-08-17 10:34:51] VERBOSE[6461] pbx.c: == Spawn extension (ext-group, 700, 14) exited non-zero on 'SIP/skype-00000003' [/code] trunk config [code] [skype] disallow=all allow=ulaw&alaw&ilbc&speex canreinvite=no context=from-trunk dtmfmode=rfc2833 host=127.0.0.1 incominglimit=1 nat=never port=5070 qualify=yes secret=someSecret type=peer username=Skype_Caller deny=0.0.0.0/0.0.0.0 permit=127.0.0.1/xxx.xxx.255.255 insecure=invite,port [/code] siptosis.cfg [code] # SipToSis configuration file # ___________________________________________ # configWatchInterval=0 logConfigFile=log.properties siptoskypeauthfile=SipToSkypeAuth.props skypetosipauthfile=SkypeToSipAuth.props SkypeOutDialingRulesFile=SkypeOutDialingRules.props SipOutDialingRulesFile=SipOutDialingRules.props ua_jar=ua.jar audioPriorityIncrease=0 jitterLevel=2 skype_connect=yes skype_audioportbase=64432 enableSkypeDtmfDetector=yes SkypeDtmfDetectorHitThreshold=90 SkypeDtmfDetectorSilenceThreshold=6 sendSipDtmfToSkype=yes sendSkypeDtmfToSip=yes inbandFullTimeDtmfDetection=yes JoinManualSkypeOutboundCallToSip=no SkypeInboundAllChannelsBusyAction=refuse SkypeInboundSipDestUnavailableAction=refuse SipInboundAllChannelsBusyAction=busy skypeclientsupportsmulticalls=no concurrentcalllimit=1 autoShutdownMinutes=0 pintimeout=8 pinretrylimit=3 destinationtimeout=12 destinationretrylimit=3 pinFile=clips/enterPin.wav destinationFile=clips/enterDest.wav dialingFile=clips/dialing.wav invalidPinFile=clips/invalidPin.wav invalidDestFile=clips/invalidDest.wav skypePinFile=clips/enterPin.wav skypeDestinationFile=clips/enterDest.wav skypeDialingFile=clips/dialing.wav skypeInvalidPinFile=clips/invalidPin.wav skypeInvalidDestFile=clips/invalidDest.wav handleEarlyMedia=yes sendSkypeIM=no skypeimmessage=You are about to receive a Skype Voice call from [callerid]. sendSkypeImDelay=2 transport_protocols=udp host_port=5070 username=SkypeCaller passwd=somePassword from_url="SkypeCaller" keepalive_time=45000 audio=yes audio_port=63200 noRtpReceivedAutoHangupSeconds=30 audio_codec=PCMU,PCMA,GSMTRI,ILBC,SPEEX audio_frame_size=240,240,160,240,160 audio_avp=-1,-1,-1,98,97 skype_audiooutgain=1,1,1,1,1 skype_audioingain=1.5,1.5,1.5,1.5,1.5 FilterParams=NONE enableSendRTPtoReceivedAddress=yes lockRtpSendAddressAfterPackets=1 dtmf2833payloadtype=101 enableSIPInbandDtmfDetector=no SipDtmfDetectorHitThreshold=30 SipDtmfDetectorSilenceThreshold=6 useViaRport=yes useViaReceived=yes sendResponseUsingOutboundProxy=no baseFailureResponse=403 skypeRefusedResponse=603 skypeFailedResponse=404 skypeUnPlacedResponse=408 skypeBusyResponse=600 TcpRxBufferSize=8192 TcpTxBufferSize=8192 RtpRxBufferSize=8192 RtpTxBufferSize=8192 is_registrar=no register_new_users=yes allowMultiContactsPerUser=no [/code] Sip to Skype and Skype to Sip Auth [code] *,*,*,calleeid *,sip:75973@127.0.0.1:5060 [/code] I suspect the problem lies with this portion of the asterisk log: [code] [2010-08-17 10:34:51] WARNING[2851] chan_sip.c: Maximum retries exceeded on transmission 238138041187@127.0.0.1 for seqno 1 (Critical Response) -- See doc/sip-retransmit.txt. [2010-08-17 10:34:51] WARNING[2851] chan_sip.c: Hanging up call 238138041187@127.0.0.1 - no reply to our critical packet (see doc/sip-retransmit.txt). [/code] Which, I think, is related to the insecure= portion of my trunk config. Unfortunately, I can't figure out how to procede. Any help would be greatly appreciated. Also, removing deny= and permit= from the trunk config didn't change the behavior. The call is refused if I leave out insecure=. Thanks]]> Wed, 18 Aug 2010 18:33:48 +0000 http://www.mhspot.com/stsforum/viewtopic.php?id=339&action=new http://www.mhspot.com/stsforum/viewtopic.php?id=338&action=new
]]> Sun, 15 Aug 2010 22:36:07 +0000 http://www.mhspot.com/stsforum/viewtopic.php?id=338&action=new http://www.mhspot.com/stsforum/viewtopic.php?id=337&action=new edit stsTrunkDef, stsTrunkPbxSipBase.conf,stsTrunkPbxExt.conf and runned trunkConfig_linux again; Result => No stsTrunk_Control script which I can use to configure channels!!! Thanks]]> Sat, 14 Aug 2010 13:28:11 +0000 http://www.mhspot.com/stsforum/viewtopic.php?id=337&action=new http://www.mhspot.com/stsforum/viewtopic.php?id=336&action=new ;tag=-3636870b4c60289d4e247597_F192.168.18.59 To: sip:5157@acme.com .... In the above 192.168.XX.XX:5070 is the IP/Port of SipToSis. I already know that mickey's skype address is mickey_mouse and have placed this information in the request uri however I can't see a way for the regex in SkypeOutDialingRules.props to make use of this information. From what I can tel the regex only has access to the user part of the To: header. (i.e. 5157) and cannot use or manipulate the user part of the request uro (i.e. mickey_mouse). Is there anyway to tweak the algorithm so that it can work on the request uri? The alternative is for me to adapt the message (with a B2B) so that I present the skype name in the To address. Thanks for your software - it works very well!]]> Thu, 12 Aug 2010 14:58:59 +0000 http://www.mhspot.com/stsforum/viewtopic.php?id=336&action=new http://www.mhspot.com/stsforum/viewtopic.php?id=326&action=new Thu, 12 Aug 2010 03:30:35 +0000 http://www.mhspot.com/stsforum/viewtopic.php?id=326&action=new http://www.mhspot.com/stsforum/viewtopic.php?id=335&action=new
]]> Tue, 10 Aug 2010 17:53:12 +0000 http://www.mhspot.com/stsforum/viewtopic.php?id=335&action=new http://www.mhspot.com/stsforum/viewtopic.php?id=334&action=new 2000,1,Dial(SIP/2000) exten => 2000,n,Hangup() exten => _7.,1,Dial(SIP/serenity/${EXTEN:1}) exten => _7.,n,Hangup() [from-skype] exten => serenity,1,Dial(SIP/2000) exten => serenity,1,Hangup() and in the sip.conf [serenity] contex=internal has changed to context=from-skype. The log messages show up in the siptosis console are 2010-08-04 14:55:23,591 callReceived - incoming Skype Call from:+13047809112 [+13047809112] status:RINGING 2010-08-04 14:55:23,666 callReceived - Command List:play:clips/invalidDest.wav 2010-08-04 14:55:24,793 skypeCallStatus[43] - INPROGRESS 2010-08-04 14:55:25,021 skypeCallStatus[43] - RINGING 2010-08-04 14:55:25,170 play:clips/invalidDest.wav 2010-08-04 14:55:29,476 skypeCallStatus[43] - Complete: FINISHED 2010-08-04 14:55:29,477 * SkypeAudioReceiver stats - packets:350 2010-08-04 14:55:29,488 * SkypeAudioSender stats - packets:0 maxLatency:0ms avgLatency:0ms 2010-08-04 14:55:29,602 INCOMING_PSTN From: +13047809112 To: NoOne CallTime: 0:04 Cost: FREE Any thoughts? B]]> Wed, 04 Aug 2010 20:48:12 +0000 http://www.mhspot.com/stsforum/viewtopic.php?id=334&action=new http://www.mhspot.com/stsforum/viewtopic.php?id=333&action=new #username=skypests #passwd=somePassword #realm=127.0.0.1 # --- end of NO registration example --- #Sample config WITH registration to GizmoProject - comment out NO registration info above first and uncomment the following#contact_url=sip:1747???????@SipToSisIpAddress:SipToSisHostPort #from_url="1747???????" #username=1747??????? #passwd=????? #realm=proxy01.sipphone.com #expires=120 #minregrenewtime=60 #regfailretrytime=15 #do_register=yes # --- end of WITH registration example --- #sample FWD reg example - note the outbound proxy #host_port=5070 #contact_url=sip:8?????@SipToSisIpAddress:SipToSisHostPort #from_url="8?????" #username=8????? #realm=fwd.pulver.com #passwd=???? #expires=240 #do_register=yes #minregrenewtime=120 #regfailretrytime=15 #outbound_proxy=fwdnat2.pulver.com:5060 # --- end of FWD registration example --- #sample pennytel reg example #host_port=5070 #contact_url=sip:8?????@SipToSisIpAddress:SipToSisHostPort #from_url="8?????" #username=8????? #realm=sip.pennytel.com #passwd=somePassword #expires=320 #do_register=yes #minregrenewtime=120 #regfailretrytime=15 # --- end of pennytel registration example --- #Sample Asterisk registration example - comment out NO registration info above first and uncomment the following. host_port=5070 contact_url=sip:serenity@192.168.1.100:5070 from_url="serenity" username=serenity realm=asterisk passwd=somePassword expires=3600 do_register=yes minregrenewtime=120 regfailretrytime=15 # --- end of Asterisk Reg example --- #Sample FreeSwitch regisration - can't use 5070 like others #host_port=5077#contact_url=sip:skypetestuser@SipToSisIpAddress:SipToSisHostPort #from_url="skypetestuser" #username=skypetestuser #realm=freeswitchDeterminedRealm #passwd=somePassword #expires=3600 #do_register=yes #minregrenewtime=160 #regfailretrytime=15 # --- end of FreeSwitch Reg example --- #do_unregister=yes #do_unregister_all=yes #keepalive_time - set to zero to disable keep alives - 0 recommended if using local SIP Server otherwise use 25000 to 45000 if not port forwarding keepalive_time=0 audio=yes #following is the SIP RTP port base - use an even port number audio_port=63200 #auto hangup after no rtp packets received for ? seconds noRtpReceivedAutoHangupSeconds=30 #only PCMU,PCMA,GSM (jmf lib),GSMTRI (tritonus libs) codecs currently supported - order by preference #You can append RW to the codec name to disable sample averaging - you will need to use filter with those - see FilterParams. #Speex doesn't not work well at all - high cpu usage and poor sound quality audio_codec=PCMU,PCMA,ILBC #PCMU/PCMA allow 160,240,320 - GSM allows 160 - ILBC allows 240, Speex allows 160, Speex16k allows 320 - need one size for each codec specified #Some ATA's don't work well with 240 for PCMU/PCMA codecs - change to 160 if needed. audio_frame_size=240,240,240 #if using dynamic payload types (speex and ilbc) you must specify the payload number (asterisk uses 98,97), if not you can remove this parameter audio_avp=-1,-1,98 #Audio volume gains - 1 for each codec - (decimal number) 1=flat (no gain), higher=louder, too high will clip or distort #volume sip->skype skype_audiooutgain=1,1,1 #volume skype->sip skype_audioingain=1.2,1.2,1.2 #Filter skype audio before being downsampled and sent to SIP device - enable a filter if using the RW codecs. #No Filtering FilterParams=NONE #RC lowpass filter #RC,delay time (lower lowers cutoff),RC constant (higher lowers cutoff) - 50,40 is a good starting point #FilterParams=RC,50,40 #FIR filter #FIR,Order (higher sharper cutoff and more cpu),window type (RECTANGULAR,HANNING,HAMMING,BLACKMAN),filter type (LP,HP,BP),minFreq,maxFreq #FilterParams=FIR,100,HANNING,LP,0,3200 #FilterParams=FIR,100,HANNING,HP,300,3200 #FilterParams=FIR,100,HANNING,BP,300,3200 #If yes, will send RTP packets to address received from the otherside # instead of what was received in the session descriptor. # This may help with one way audio problems. enableSendRTPtoReceivedAddress=yes #works with above setting - sending of rtpPackets can be redirected until receiving this number of packets. After that the address is locked. lockRtpSendAddressAfterPackets=1 #Set to -1 to disable rfc2833 some providers use 96 most use 101 dtmf2833payloadtype=101 #Use these for SIP INFO msg support - first is the most common type #dtmfinfotype=application/dtmf-relay #dtmfinfotype=application/dtmf #Use only if rfc2833 and INFO are not supported - uses more cpu enableSIPInbandDtmfDetector=no SIPInbandDtmfDetectorGain=1 SipDtmfDetectorHitThreshold=30 SipDtmfDetectorSilenceThreshold=40 #shutoff detector after time has elasped autoDisableSipInbandDtmfDetectorSeconds=80 #params to control sip response address handling useViaRport=yes useViaReceived=yes #send all responses using outbound proxy - outbound proxy must be set up sendResponseUsingOutboundProxy=no #sip response for any uncovered reason (possibly no skype credit) baseFailureResponse=403 #sip response if remote skype user refused call skypeRefusedResponse=603 #sip response if skype call failed (possibly no skype credit) skypeFailedResponse=408 #sip response if invalid skype user or number skypeInvalidDestinationResponse=404 #sip response if skype returned unplaced status skypeUnPlacedResponse=408 #sip response if called party is busy skypeBusyResponse=600 #network buffers for skype api audio transport (0=leave at OS default) TcpRxBufferSize=8192 TcpTxBufferSize=8192 #network buffers for RTP audio transport (0=leave at OS default) RtpRxBufferSize=8192 RtpTxBufferSize=8192 #TOS flags (recommended 0x10 or 0x18) -1 to use platform default # On windows make sure DWORD HKEY_LOCAL_MACHINE\System\CurrentControlSet\Services\Tcpip\Parameters\DisableUserTOSSetting=0 Does nothing on Vista and later RtpTosFlags=0x10 #*** register server settings below *** not required if registration is not needed for phone #set to yes to turn on server registrar or no to disable is_registrar=yes #set to yes to allow register of users not already in registar database (users.db) register_new_users=yes #set to domains of server - see mjsip doc #domain_names=192.168.0.4 somedomain.com allowMultiContactsPerUser=no #enable if your ATA/SIP device does not send the port in the request line (FritzBox and others) domain_port_any=yes #voicemail clips vmNoMessagesClip=clips/vmpnomessages.wav vmPlayingClip=clips/vmpplayingmessages.wav vmEndOfMessageClip=clips/vmpendofmessageprompt.wav vmMessageDeletedClip=clips/vmpmessagedeleted.wav vmMessageDeleteAllClip=clips/vmpmessagedeleteall.wav vmNoMoreMessagesClip=clips/vmpnomoremessages.wav vmRecordingBeepClip=clips/vmprecordingbeep.wav sip.conf [general] bindaddr=0.0.0.0 context=default allowoverlap=no srvlookup=yes disallow=all allow=ulaw allow=alaw allow=g726 register => serenity:skypetest@192.168.1.100/serenity [serenity] username=serenity type=friend context=internal secret=someSecret host=dynamic nat=no dtmfmode=auto canreinvite=no defaultip=192.168.1.100 qualify=yes incominglimit=1 outgoinglimit=1 call-limit=1 busylevel=1 [2000] type=friend context=internal secret=someSecret host=dynamic qualify=yes extensions.conf [internal] exten => _7.,1,Dial(SIP/serenity/${EXTEN:1}) exten => _7.,n,Hangup() [from-skype] exten => _00X.,1,Answer() exten => _00X.,n,Dial(SIP/2000, 20) exten => _00X.,n,VoiceMail(2000, u) exten => _00X.,n,Hangup() I can't figure out where the loop is. Hopefully you can help. B]]> Wed, 04 Aug 2010 17:40:06 +0000 http://www.mhspot.com/stsforum/viewtopic.php?id=333&action=new