Входящие звонки на Cisco 7911 и Asterisk [закрыт]

Добрый день Подскажите куда копать. Есть Asterisk (сборка на базе Elastix) Софтфоны работают без проблем, пытаюсь подружить с ним телефоны Cisco IP Phone 7911. Прошил под SIP, подправил SEPxxxxx.xml. Cisco 7911 регистрируется на asterisk, исходящие звонки идут без проблем, а вот с входящими засада. Пытаюсь позвонить на Cisco 7911? а в ответ отбой. С asterisk недавно работаю, сильно не ругайтесь
конфиги и лог прикладываю

sip.conf

[1010] deny=0.0.0.0/0.0.0.0 secret=1010 dtmfmode=rfc2833 canreinvite=no context=from-internal host=dynamic type=friend nat=no port=5060 qualify=yes callgroup= pickupgroup= dial=SIP/1010 mailbox=1010@device permit=0.0.0.0/0.0.0.0 callerid=device <1010> callcounter=yes faxdetect=no 

SEPxxxxxxx.xml

<device xsi:type="axl:XIPPhone" ctiid="1566023366"> <deviceProtocol>SIP</deviceProtocol> <sshUserId>cisco</sshUserId> <sshPassword>cisco</sshPassword> <devicePool> <dateTimeSetting> <dateTemplate>D-M-YA</dateTemplate> <timeZone>Russian Standard/Daylight Time </timeZone> <ntp> <name>10.122.122.1</name> <ntpMode>Unicast</ntpMode> </ntp> </dateTimeSetting> <callManagerGroup> <members> <member priority="0"> <callManager> <ports> <ethernetPhonePort>2000</ethernetPhonePort> <sipPort>5060</sipPort> <securedSipPort>5061</securedSipPort> </ports> <processNodeName>10.121.0.95</processNodeName> </callManager> </member> </members> </callManagerGroup> </devicePool> <sipProfile> <sipProxies> <backupProxy></backupProxy> <backupProxyPort></backupProxyPort> <emergencyProxy></emergencyProxy> <emergencyProxyPort></emergencyProxyPort> <outboundProxy></outboundProxy> <outboundProxyPort></outboundProxyPort> <registerWithProxy>false</registerWithProxy> </sipProxies> <sipCallFeatures> <cnfJoinEnabled>true</cnfJoinEnabled> <callForwardURI>x--serviceuri-cfwdall</callForwardURI> <callPickupURI>x-cisco-serviceuri-pickup</callPickupURI> <callPickupListURI>x-cisco-serviceuri-opickup</callPickupListURI> <callPickupGroupURI>x-cisco-serviceuri-gpickup</callPickupGroupURI> <meetMeServiceURI>x-cisco-serviceuri-meetme</meetMeServiceURI> <abbreviatedDialURI>x-cisco-serviceuri-abbrdial</abbreviatedDialURI> <rfc2543Hold>false</rfc2543Hold> <callHoldRingback>2</callHoldRingback> <localCfwdEnable>true</localCfwdEnable> <semiAttendedTransfer>true</semiAttendedTransfer> <anonymousCallBlock>2</anonymousCallBlock> <callerIdBlocking>2</callerIdBlocking> <dndControl>0</dndControl> <remoteCcEnable>true</remoteCcEnable> </sipCallFeatures> <sipStack> <sipInviteRetx>6</sipInviteRetx> <sipRetx>10</sipRetx> <timerInviteExpires>180</timerInviteExpires> <timerRegisterExpires>3600</timerRegisterExpires> <timerRegisterDelta>5</timerRegisterDelta> <timerKeepAliveExpires>120</timerKeepAliveExpires> <timerSubscribeExpires>120</timerSubscribeExpires> <timerSubscribeDelta>5</timerSubscribeDelta> <timerT1>500</timerT1> <timerT2>4000</timerT2> <maxRedirects>70</maxRedirects> <remotePartyID>false</remotePartyID> <userInfo>None</userInfo> </sipStack> <autoAnswerTimer>1</autoAnswerTimer> <autoAnswerAltBehavior>false</autoAnswerAltBehavior> <autoAnswerOverride>true</autoAnswerOverride> <transferOnhookEnabled>false</transferOnhookEnabled> <enableVad>false</enableVad> <dtmfAvtPayload>101</dtmfAvtPayload> <dtmfDbLevel>3</dtmfDbLevel> <dtmfOutofBand>avt</dtmfOutofBand> <alwaysUsePrimeLine>false</alwaysUsePrimeLine> <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail> <kpml>3</kpml> <phoneLabel>1010</phoneLabel> <stutterMsgWaiting>1</stutterMsgWaiting> <callStats>false</callStats> <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts> <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig> <startMediaPort>16384</startMediaPort> <stopMediaPort>32766</stopMediaPort> <natEnabled>false</natEnabled> <natAddress></natAddress> <sipLines> <line button="1"> <featureID>9</featureID> <featureLabel>1010</featureLabel> <proxy>10.121.0.95</proxy> <port>5060</port> <name>1010</name> <displayName>1010</displayName> <autoAnswer> <autoAnswerEnabled>2</autoAnswerEnabled> </autoAnswer> <callWaiting>3</callWaiting> <authName>1010</authName> <authPassword>1010</authPassword> <sharedLine>false</sharedLine> <messageWaitingLampPolicy>1</messageWaitingLampPolicy> <messagesNumber>999</messagesNumber> <ringSettingIdle>4</ringSettingIdle> <ringSettingActive>5</ringSettingActive> <contact>1010</contact> <forwardCallInfoDisplay> <callerName>true</callerName> <callerNumber>true</callerNumber> <redirectedNumber>false</redirectedNumber> <dialedNumber>true</dialedNumber> </forwardCallInfoDisplay> </line> </sipLines> <voipControlPort>5060</voipControlPort> <dscpForAudio>184</dscpForAudio> <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy> <dialTemplate>dialplan.xml</dialTemplate> </sipProfile> <commonProfile> <phonePassword></phonePassword> <backgroundImageAccess>true</backgroundImageAccess> <callLogBlfEnabled>2</callLogBlfEnabled> </commonProfile> <loadInformation>SIP11.8-4-2S</loadInformation> <vendorConfig> <disableSpeaker>false</disableSpeaker> <disableSpeakerAndHeadset>false</disableSpeakerAndHeadset> <pcPort>0</pcPort> <settingsAccess>1</settingsAccess> <garp>0</garp> <voiceVlanAccess>0</voiceVlanAccess> <videoCapability>0</videoCapability> <autoSelectLineEnable>0</autoSelectLineEnable> <webAccess>1</webAccess> <daysDisplayNotActive>1,2,3,4,5,6,7</daysDisplayNotActive> <displayOnTime>00:00</displayOnTime> <displayOnDuration>00:00</displayOnDuration> <displayIdleTimeout>00:00</displayIdleTimeout> <spanToPCPort>1</spanToPCPort> <loggingDisplay>1</loggingDisplay> <loadServer></loadServer> </vendorConfig> <versionStamp>1143565489-a3cbf294-7526-4c29-8791-c4fce4ce4c37</versionStamp> <networkLocale>New_Zealand</networkLocale> <networkLocaleInfo> <name>New_Zealand</name> <version>5.0(2)</version> </networkLocaleInfo> <deviceSecurityMode>1</deviceSecurityMode> <authenticationURL>http://www/ipphone/authenticate.php</authenticationURL> <directoryURL>http://www/ipphone/directory.xml</directoryURL> <idleURL></idleURL> <informationURL>http://www/ipphone/GetTelecasterHelpText.jsp</informationURL> <messagesURL></messagesURL> <proxyServerURL>proxy:3128</proxyServerURL> <servicesURL>http://www/ipphone/services.xml</servicesURL> <dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig> <dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices> <dscpForCm2Dvce>96</dscpForCm2Dvce> <transportLayerProtocol>4</transportLayerProtocol> <capfAuthMode>0</capfAuthMode> <capfList> <capf> <phonePort>3804</phonePort> </capf> </capfList> <certHash></certHash> <encrConfig>false</encrConfig> </device> 

Дебаг

<--- SIP read from UDP:10.121.80.254:50771 ---> INVITE sip:1010@10.121.0.95:5060 SIP/2.0 Via: SIP/2.0/UDP 10.121.80.254:50771;branch=z9hG4bK-d8754z-f44c4101c6065235-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1000@10.121.80.254:50771;rinstance=240eebe44b7bff8e> To: <sip:1010@10.121.0.95:5060> From: "1000"<sip:1000@10.121.0.95:5060>;tag=2f428621 Call-ID: MzExYjMxZDg5NDUwODQ2MzI2N2Q5NzY2ZTdkZjgzYzk. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Content-Type: application/sdp Supported: replaces User-Agent: 3CXPhone 5.0.14900.0 Content-Length: 408 v=0 o=3cxVCE 110080365 212704350 IN IP4 10.121.80.254 s=3cxVCE Audio Call c=IN IP4 10.121.80.254 t=0 0 m=audio 40048 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv m=video 40010 RTP/AVP 34 c=IN IP4 10.121.80.254 a=rtpmap:34 H263/90000 a=fmtp:34 CIF4=1;CIF=1;QCIF=1;SQCIF=1; a=sendrecv <-------------> --- (13 headers 18 lines) --- Sending to 10.121.80.254:50771 (NAT) Using INVITE request as basis request - MzExYjMxZDg5NDUwODQ2MzI2N2Q5NzY2ZTdkZjgzYzk. Found peer '1000' for '1000' from 10.121.80.254:50771 <--- Reliably Transmitting (NAT) to 10.121.80.254:50771 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.121.80.254:50771;branch=z9hG4bK-d8754z-f44c4101c6065235-1---d8754z-;received=10.121.80.254;rport=50771 From: "1000"<sip:1000@10.121.0.95:5060>;tag=2f428621 To: <sip:1010@10.121.0.95:5060>;tag=as6faacb77 Call-ID: MzExYjMxZDg5NDUwODQ2MzI2N2Q5NzY2ZTdkZjgzYzk. CSeq: 1 INVITE Server: FPBX-2.8.1(1.8.11.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="30e42dba" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'MzExYjMxZDg5NDUwODQ2MzI2N2Q5NzY2ZTdkZjgzYzk.' in 6528 ms (Method: INVITE) <--- SIP read from UDP:10.121.80.254:50771 ---> ACK sip:1010@10.121.0.95:5060 SIP/2.0 Via: SIP/2.0/UDP 10.121.80.254:50771;branch=z9hG4bK-d8754z-f44c4101c6065235-1---d8754z-;rport Max-Forwards: 70 To: <sip:1010@10.121.0.95:5060>;tag=as6faacb77 From: "1000"<sip:1000@10.121.0.95:5060>;tag=2f428621 Call-ID: MzExYjMxZDg5NDUwODQ2MzI2N2Q5NzY2ZTdkZjgzYzk. CSeq: 1 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:10.121.80.254:50771 ---> INVITE sip:1010@10.121.0.95:5060 SIP/2.0 Via: SIP/2.0/UDP 10.121.80.254:50771;branch=z9hG4bK-d8754z-3f04f26bca759626-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1000@10.121.80.254:50771;rinstance=240eebe44b7bff8e> To: <sip:1010@10.121.0.95:5060> From: "1000"<sip:1000@10.121.0.95:5060>;tag=2f428621 Call-ID: MzExYjMxZDg5NDUwODQ2MzI2N2Q5NzY2ZTdkZjgzYzk. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Content-Type: application/sdp Supported: replaces User-Agent: 3CXPhone 5.0.14900.0 Authorization: Digest username="1000",realm="asterisk",nonce="30e42dba",uri="sip:1010@10.121.0.95:5060",response="fcb60ae429ecc89755d7f224475d38a2",algorithm=MD5 Content-Length: 408 v=0 o=3cxVCE 110080365 212704350 IN IP4 10.121.80.254 s=3cxVCE Audio Call c=IN IP4 10.121.80.254 t=0 0 m=audio 40048 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv m=video 40010 RTP/AVP 34 c=IN IP4 10.121.80.254 a=rtpmap:34 H263/90000 a=fmtp:34 CIF4=1;CIF=1;QCIF=1;SQCIF=1; a=sendrecv <-------------> --- (14 headers 18 lines) --- Sending to 10.121.80.254:50771 (NAT) Using INVITE request as basis request - MzExYjMxZDg5NDUwODQ2MzI2N2Q5NzY2ZTdkZjgzYzk. Found peer '1000' for '1000' from 10.121.80.254:50771 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format GSM for ID 3 Found audio description format telephone-event for ID 101 Found RTP video format 34 Found video description format H263 for ID 34 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x80000 (h263)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.121.80.254:40048 Looking for 1010 in from-internal (domain 10.121.0.95) list_route: hop: <sip:1000@10.121.80.254:50771;rinstance=240eebe44b7bff8e> <--- Transmitting (NAT) to 10.121.80.254:50771 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.121.80.254:50771;branch=z9hG4bK-d8754z-3f04f26bca759626-1---d8754z-;received=10.121.80.254;rport=50771 From: "1000"<sip:1000@10.121.0.95:5060>;tag=2f428621 To: <sip:1010@10.121.0.95:5060> Call-ID: MzExYjMxZDg5NDUwODQ2MzI2N2Q5NzY2ZTdkZjgzYzk. CSeq: 2 INVITE Server: FPBX-2.8.1(1.8.11.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:1010@10.121.0.95:5060> Content-Length: 0 <------------> -- Executing [1010@from-internal:1] Macro("SIP/1000-0000001f", "exten-vm,novm,1010") in new stack -- Executing [s@macro-exten-vm:1] Macro("SIP/1000-0000001f", "user-callerid,") in new stack -- Executing [s@macro-user-callerid:1] Set("SIP/1000-0000001f", "AMPUSER=1000") in new stack -- Executing [s@macro-user-callerid:2] GotoIf("SIP/1000-0000001f", "0?report") in new stack -- Executing [s@macro-user-callerid:3] ExecIf("SIP/1000-0000001f", "1?Set(REALCALLERIDNUM=1000)") in new stack -- Executing [s@macro-user-callerid:4] Set("SIP/1000-0000001f", "AMPUSER=1000") in new stack -- Executing [s@macro-user-callerid:5] Set("SIP/1000-0000001f", "AMPUSERCIDNAME=Admin") in new stack -- Executing [s@macro-user-callerid:6] GotoIf("SIP/1000-0000001f", "0?report") in new stack -- Executing [s@macro-user-callerid:7] Set("SIP/1000-0000001f", "AMPUSERCID=1000") in new stack -- Executing [s@macro-user-callerid:8] Set("SIP/1000-0000001f", "CALLERID(all)="Admin" <1000>") in new stack -- Executing [s@macro-user-callerid:9] ExecIf("SIP/1000-0000001f", "0?Set(CHANNEL(language)=)") in new stack -- Executing [s@macro-user-callerid:10] GotoIf("SIP/1000-0000001f", "0?continue") in new stack -- Executing [s@macro-user-callerid:11] Set("SIP/1000-0000001f", "__TTL=64") in new stack -- Executing [s@macro-user-callerid:12] GotoIf("SIP/1000-0000001f", "1?continue") in new stack -- Goto (macro-user-callerid,s,19) -- Executing [s@macro-user-callerid:19] Set("SIP/1000-0000001f", "CALLERID(number)=1000") in new stack -- Executing [s@macro-user-callerid:20] Set("SIP/1000-0000001f", "CALLERID(name)=Admin") in new stack -- Executing [s@macro-user-callerid:21] NoOp("SIP/1000-0000001f", "Using CallerID "Admin" <1000>") in new stack -- Executing [s@macro-exten-vm:2] Set("SIP/1000-0000001f", "RingGroupMethod=none") in new stack -- Executing [s@macro-exten-vm:3] Set("SIP/1000-0000001f", "VMBOX=novm") in new stack -- Executing [s@macro-exten-vm:4] Set("SIP/1000-0000001f", "__EXTTOCALL=1010") in new stack -- Executing [s@macro-exten-vm:5] Set("SIP/1000-0000001f", "CFUEXT=") in new stack -- Executing [s@macro-exten-vm:6] Set("SIP/1000-0000001f", "CFBEXT=") in new stack -- Executing [s@macro-exten-vm:7] Set("SIP/1000-0000001f", "RT=""") in new stack -- Executing [s@macro-exten-vm:8] Macro("SIP/1000-0000001f", "record-enable,1010,IN") in new stack -- Executing [s@macro-record-enable:1] GotoIf("SIP/1000-0000001f", "1?check") in new stack -- Goto (macro-record-enable,s,4) -- Executing [s@macro-record-enable:4] ExecIf("SIP/1000-0000001f", "0?MacroExit()") in new stack -- Executing [s@macro-record-enable:5] GotoIf("SIP/1000-0000001f", "0?Group:OUT") in new stack -- Goto (macro-record-enable,s,15) -- Executing [s@macro-record-enable:15] GotoIf("SIP/1000-0000001f", "1?IN") in new stack -- Goto (macro-record-enable,s,20) -- Executing [s@macro-record-enable:20] ExecIf("SIP/1000-0000001f", "0?MacroExit()") in new stack -- Executing [s@macro-record-enable:21] NoOp("SIP/1000-0000001f", "Recording enable for 1010") in new stack -- Executing [s@macro-record-enable:22] Set("SIP/1000-0000001f", "CALLFILENAME=20120425-141731-1335349051.39") in new stack -- Executing [s@macro-record-enable:23] MixMonitor("SIP/1000-0000001f", "20120425-141731-1335349051.39.wav,,") in new stack -- Executing [s@macro-record-enable:24] Set("SIP/1000-0000001f", "CDR(userfield)=audio:20120425-141731-1335349051.39.wav") in new stack -- Executing [s@macro-record-enable:25] MacroExit("SIP/1000-0000001f", "") in new stack -- Executing [s@macro-exten-vm:9] Macro("SIP/1000-0000001f", "dial-one,"",tr,1010") in new stack -- Executing [s@macro-dial-one:1] Set("SIP/1000-0000001f", "DEXTEN=1010") in new stack -- Executing [s@macro-dial-one:2] Set("SIP/1000-0000001f", "DIALSTATUS_CW=") in new stack -- Executing [s@macro-dial-one:3] GosubIf("SIP/1000-0000001f", "0?screen,1") in new stack -- Executing [s@macro-dial-one:4] GosubIf("SIP/1000-0000001f", "0?cf,1") in new stack -- Executing [s@macro-dial-one:5] GotoIf("SIP/1000-0000001f", "1?skip1") in new stack -- Goto (macro-dial-one,s,8) -- Executing [s@macro-dial-one:8] GotoIf("SIP/1000-0000001f", "0?nodial") in new stack -- Executing [s@macro-dial-one:9] GotoIf("SIP/1000-0000001f", "0?continue") in new stack -- Executing [s@macro-dial-one:10] Set("SIP/1000-0000001f", "EXTHASCW=") in new stack -- Executing [s@macro-dial-one:11] GotoIf("SIP/1000-0000001f", "1?next1:cwinusebusy") in new stack -- Goto (macro-dial-one,s,12) -- Executing [s@macro-dial-one:12] GotoIf("SIP/1000-0000001f", "0?docfu:skip3") in new stack -- Goto (macro-dial-one,s,16) -- Executing [s@macro-dial-one:16] GotoIf("SIP/1000-0000001f", "1?next2:continue") in new stack -- Goto (macro-dial-one,s,17) -- Executing [s@macro-dial-one:17] GotoIf("SIP/1000-0000001f", "1?continue") in new stack -- Goto (macro-dial-one,s,25) -- Executing [s@macro-dial-one:25] GotoIf("SIP/1000-0000001f", "0?nodial") in new stack -- Executing [s@macro-dial-one:26] GosubIf("SIP/1000-0000001f", "1?dstring,1:dlocal,1") in new stack -- Executing [dstring@macro-dial-one:1] Set("SIP/1000-0000001f", "DSTRING=") in new stack -- Executing [dstring@macro-dial-one:2] Set("SIP/1000-0000001f", "DEVICES=1010") in new stack -- Executing [dstring@macro-dial-one:3] ExecIf("SIP/1000-0000001f", "0?Return()") in new stack -- Executing [dstring@macro-dial-one:4] ExecIf("SIP/1000-0000001f", "0?Set(DEVICES=010)") in new stack -- Executing [dstring@macro-dial-one:5] Set("SIP/1000-0000001f", "LOOPCNT=1") in new stack -- Executing [dstring@macro-dial-one:6] Set("SIP/1000-0000001f", "ITER=1") in new stack -- Executing [dstring@macro-dial-one:7] Set("SIP/1000-0000001f", "THISDIAL=SIP/1010") in new stack -- Executing [dstring@macro-dial-one:8] GosubIf("SIP/1000-0000001f", "1?zap2dahdi,1") in new stack == Begin MixMonitor Recording SIP/1000-0000001f -- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/1000-0000001f", "0?Return()") in new stack -- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/1000-0000001f", "NEWDIAL=") in new stack -- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/1000-0000001f", "LOOPCNT2=1") in new stack -- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/1000-0000001f", "ITER2=1") in new stack -- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/1000-0000001f", "THISPART2=SIP/1010") in new stack -- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/1000-0000001f", "0?Set(THISPART2=DAHDI/1010)") in new stack -- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/1000-0000001f", "NEWDIAL=SIP/1010&") in new stack -- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/1000-0000001f", "ITER2=2") in new stack -- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/1000-0000001f", "0?begin2") in new stack -- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/1000-0000001f", "THISDIAL=SIP/1010") in new stack -- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/1000-0000001f", "") in new stack -- Executing [dstring@macro-dial-one:9] Set("SIP/1000-0000001f", "DSTRING=SIP/1010&") in new stack -- Executing [dstring@macro-dial-one:10] Set("SIP/1000-0000001f", "ITER=2") in new stack -- Executing [dstring@macro-dial-one:11] GotoIf("SIP/1000-0000001f", "0?begin") in new stack -- Executing [dstring@macro-dial-one:12] Set("SIP/1000-0000001f", "DSTRING=SIP/1010") in new stack -- Executing [dstring@macro-dial-one:13] Return("SIP/1000-0000001f", "") in new stack -- Executing [s@macro-dial-one:27] GotoIf("SIP/1000-0000001f", "0?nodial") in new stack -- Executing [s@macro-dial-one:28] GotoIf("SIP/1000-0000001f", "1?skiptrace") in new stack -- Goto (macro-dial-one,s,30) -- Executing [s@macro-dial-one:30] Set("SIP/1000-0000001f", "D_OPTIONS=tr") in new stack -- Executing [s@macro-dial-one:31] ExecIf("SIP/1000-0000001f", "0?SIPAddHeader(Alert-Info: )") in new stack -- Executing [s@macro-dial-one:32] ExecIf("SIP/1000-0000001f", "0?SIPAddHeader()") in new stack -- Executing [s@macro-dial-one:33] ExecIf("SIP/1000-0000001f", "0?Set(CHANNEL(musicclass)=)") in new stack -- Executing [s@macro-dial-one:34] GosubIf("SIP/1000-0000001f", "0?qwait,1") in new stack -- Executing [s@macro-dial-one:35] Set("SIP/1000-0000001f", "__CWIGNORE=") in new stack -- Executing [s@macro-dial-one:36] Set("SIP/1000-0000001f", "__KEEPCID=TRUE") in new stack -- Executing [s@macro-dial-one:37] Dial("SIP/1000-0000001f", "SIP/1010,"",tr") in new stack Really destroying SIP dialog '222511566e752f0b0391f8f7250ab533@127.0.0.1:5060' Method: INVITE == Everyone is busy/congested at this time (1:0/0/1) -- Executing [s@macro-dial-one:38] ExecIf("SIP/1000-0000001f", "0?Set(DIALSTATUS=)") in new stack -- Executing [s@macro-dial-one:39] GosubIf("SIP/1000-0000001f", "0?s-CHANUNAVAIL,1") in new stack -- Executing [s@macro-dial-one:40] MacroExit("SIP/1000-0000001f", "") in new stack -- Executing [s@macro-exten-vm:10] GotoIf("SIP/1000-0000001f", "0?exit") in new stack -- Executing [s@macro-exten-vm:11] Set("SIP/1000-0000001f", "SV_DIALSTATUS=CHANUNAVAIL") in new stack -- Executing [s@macro-exten-vm:12] GosubIf("SIP/1000-0000001f", "0?docfu,1") in new stack -- Executing [s@macro-exten-vm:13] GosubIf("SIP/1000-0000001f", "0?docfb,1") in new stack -- Executing [s@macro-exten-vm:14] Set("SIP/1000-0000001f", "DIALSTATUS=CHANUNAVAIL") in new stack -- Executing [s@macro-exten-vm:15] NoOp("SIP/1000-0000001f", "Voicemail is 'novm'") in new stack -- Executing [s@macro-exten-vm:16] GotoIf("SIP/1000-0000001f", "1?s-CHANUNAVAIL,1") in new stack -- Goto (macro-exten-vm,s-CHANUNAVAIL,1) -- Executing [s-CHANUNAVAIL@macro-exten-vm:1] NoOp("SIP/1000-0000001f", "IVR_RETVM: IVR_CONTEXT: ") in new stack -- Executing [s-CHANUNAVAIL@macro-exten-vm:2] GotoIf("SIP/1000-0000001f", "0?exit,1") in new stack -- Executing [s-CHANUNAVAIL@macro-exten-vm:3] PlayTones("SIP/1000-0000001f", "congestion") in new stack -- Executing [s-CHANUNAVAIL@macro-exten-vm:4] Congestion("SIP/1000-0000001f", "10") in new stack <--- Reliably Transmitting (NAT) to 10.121.80.254:50771 ---> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 10.121.80.254:50771;branch=z9hG4bK-d8754z-3f04f26bca759626-1---d8754z-;received=10.121.80.254;rport=50771 From: "1000"<sip:1000@10.121.0.95:5060>;tag=2f428621 To: <sip:1010@10.121.0.95:5060>;tag=as3cd55a71 Call-ID: MzExYjMxZDg5NDUwODQ2MzI2N2Q5NzY2ZTdkZjgzYzk. CSeq: 2 INVITE Server: FPBX-2.8.1(1.8.11.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-Asterisk-HangupCause: Unknown X-Asterisk-HangupCauseCode: 20 Content-Length: 0 <------------> == Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 4) exited non-zero on 'SIP/1000-0000001f' in macro 'exten-vm' == Spawn extension (from-internal, 1010, 1) exited non-zero on 'SIP/1000-0000001f' -- Executing [h@from-internal:1] Macro("SIP/1000-0000001f", "hangupcall") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("SIP/1000-0000001f", "0?endmixmoncheck") in new stack <--- SIP read from UDP:10.121.80.254:50771 ---> ACK sip:1010@10.121.0.95:5060 SIP/2.0 Via: SIP/2.0/UDP 10.121.80.254:50771;branch=z9hG4bK-d8754z-3f04f26bca759626-1---d8754z-;rport Max-Forwards: 70 To: <sip:1010@10.121.0.95:5060>;tag=as3cd55a71 From: "1000"<sip:1000@10.121.0.95:5060>;tag=2f428621 Call-ID: MzExYjMxZDg5NDUwODQ2MzI2N2Q5NzY2ZTdkZjgzYzk. CSeq: 2 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- -- Executing [s@macro-hangupcall:2] Set("SIP/1000-0000001f", "MIXMON_CALLFILENAME=/var/spool/asterisk/monitor/20120425-141731-1335349051.39.wav") in new stack -- Executing [s@macro-hangupcall:3] GotoIf("SIP/1000-0000001f", "1?defaultmixmondir") in new stack -- Goto (macro-hangupcall,s,5) -- Executing [s@macro-hangupcall:5] System("SIP/1000-0000001f", "test -e /var/spool/asterisk/monitor/20120425-141731-1335349051.39.wav") in new stack -- Executing [s@macro-hangupcall:6] NoOp("SIP/1000-0000001f", "SYSTEMSTATUS = APPERROR") in new stack -- Executing [s@macro-hangupcall:7] GotoIf("SIP/1000-0000001f", "0?endmixmoncheck") in new stack -- Executing [s@macro-hangupcall:8] Set("SIP/1000-0000001f", "CDR(userfield)=") in new stack -- Executing [s@macro-hangupcall:9] NoOp("SIP/1000-0000001f", "End of MIXMON check") in new stack -- Executing [s@macro-hangupcall:10] GotoIf("SIP/1000-0000001f", "1?nomeetmemon") in new stack -- Goto (macro-hangupcall,s,15) -- Executing [s@macro-hangupcall:15] NoOp("SIP/1000-0000001f", "MEETME_RECORDINGFILE=") in new stack -- Executing [s@macro-hangupcall:16] GotoIf("SIP/1000-0000001f", "1?noautomon") in new stack -- Goto (macro-hangupcall,s,18) -- Executing [s@macro-hangupcall:18] NoOp("SIP/1000-0000001f", "TOUCH_MONITOR_OUTPUT=") in new stack -- Executing [s@macro-hangupcall:19] GotoIf("SIP/1000-0000001f", "1?noautomon2") in new stack -- Goto (macro-hangupcall,s,25) -- Executing [s@macro-hangupcall:25] NoOp("SIP/1000-0000001f", "MONITOR_FILENAME=") in new stack -- Executing [s@macro-hangupcall:26] GotoIf("SIP/1000-0000001f", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,29) -- Executing [s@macro-hangupcall:29] GotoIf("SIP/1000-0000001f", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,32) -- Executing [s@macro-hangupcall:32] GotoIf("SIP/1000-0000001f", "1?theend") in new stack -- Goto (macro-hangupcall,s,34) -- Executing [s@macro-hangupcall:34] Hangup("SIP/1000-0000001f", "") in new stack == Spawn extension (macro-hangupcall, s, 34) exited non-zero on 'SIP/1000-0000001f' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/1000-0000001f' == End MixMonitor Recording SIP/1000-0000001f Really destroying SIP dialog 'MzExYjMxZDg5NDUwODQ2MzI2N2Q5NzY2ZTdkZjgzYzk.' Method: ACK