Входящие звонки на Cisco 7911 и Asterisk [закрыт]

Question

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

Answer 1

sip show peer 1010


  * Name       : 1010
  Secret       : <Set>
  MD5Secret    : <Not set>
  Remote Secret: <Not set>
  Context      : from-internal
  Subscr.Cont. : <Not set>
  Language     : ru
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    : 
  Pickupgroup  : 
  MOH Suggest  : 
  Mailbox      : 1010@device
  VM Extension : *97
  LastMsgsSent : 32767/65535
  Call limit   : 2147483647
  Max forwards : 0
  Dynamic      : Yes
  Callerid     : "device" <1010>
  MaxCallBR    : 384 kbps
  Expire       : -1
  Insecure     : no
  Force rport  : No
  ACL          : Yes
  DirectMedACL : No
  T.38 support : No
  T.38 EC mode : Unknown
  T.38 MaxDtgrm: -1
  DirectMedia  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Text Support : No
  Ign SDP ver  : No
  Trust RPID   : No
  Send RPID    : No
  Subscriptions: Yes
  Overlap dial : Yes
  DTMFmode     : rfc2833
  Timer T1     : 500
  Timer B      : 32000
  ToHost       : 
  Addr->IP     : (null)
  Defaddr->IP  : (null)
  Prim.Transp. : UDP
  Allowed.Trsp : UDP
  Def. Username: 1010
  SIP Options  : join norefersub replaces replace 
  Codecs       : 0xe (gsm|ulaw|alaw)
  Codec Order  : (ulaw:20,alaw:20,gsm:20)
  Auto-Framing :  No 
  Status       : Unmonitored
  Useragent    : 3CXPhone 5.0.14900.0
  Reg. Contact : sip:1010@10.121.80.254:50675;rinstance=c21b4144ac0726ae
  Qualify Freq : 60000 ms
  Sess-Timers  : Accept
  Sess-Refresh : uas
  Sess-Expires : 1800 secs
  Min-Sess     : 90 secs
  RTP Engine   : asterisk
  Parkinglot   : 
  Use Reason   : No
  Encryption   : No

Answer 2

0

правильность конфига для данных телефонов зависит от версии прошивки. сам вчера 6 часов потратил на телефон.

после люого изменения конфига идите смотрите статус и проверяйте что он вообще конфиг принял.

по теме - почему нет дебага для телефона на который звоните? может он вообещ не зарегистрирован и туда не звонит?

ссылка удалить спам редактировать

ответил 2012-04-25 23:06:39 +0400

meral

23347 ● 24 ● 20 ● 177
http://pro-sip.net/

Comments

а как сделать дебаг для телефона (не судите строго, я новичек)

CTV ( 2012-04-26 08:39:01 +0400 )редактировать

ну в 7961 в меню можно посмотреть состояние.7911 не видел вживую. еще можно зайти по ссш на телефон.

meral ( 2012-04-26 13:33:25 +0400 )редактировать

Answer 3

0

Ура заработало Выставил <registerwithproxy>true</registerwithproxy> в true Хотя везде читал что надо выставлять false

ссылка удалить спам редактировать

ответил 2012-04-26 09:13:21 +0400

CTV

1 ● 2 ● 2

Comments

везде напичано true. гед вы нашли конфиг без регистрации я не заню вообще. и не отвечайте сами себе.пишите это в КОМЕНТАРИИ.

meral ( 2012-04-26 13:34:50 +0400 )редактировать

Answer 4

sip show peer 1010

http://10.121.0.95/

вот отличия diff -abB ./SEPxxxxx.xml SEPXXXXXXXXXX.cnf.xml

  diff -abB ./SEPxxxxx.xml.txt SEPXXXXXXXXXX.cnf.xml
  <registerWithProxy>true</registerWithProxy>
  <callForwardURI>x-cisco-serviceuri-cfwdall</callForwardURI>
  <remotePartyID>true</remotePartyID>
  <preferredCodec>none</preferredCodec>
  <stutterMsgWaiting>0</stutterMsgWaiting>
  <softKeyFile>softkey.xml</softKeyFile>
  <callLogBlfEnabled>0</callLogBlfEnabled>
  <loadInformation>SIP11.8-5-2S</loadInformation>
  <forwardingDelay>1</forwardingDelay>
  <voiceVlanAccess>1</voiceVlanAccess>
  <webAccess>0</webAccess>
  <loggingDisplay>2</loggingDisplay>
  <loadServer>*.*.*.*</loadServer>
  <recordingTone>0</recordingTone>
  <recordingToneLocalVolume>100</recordingToneLocalVolume>
  <recordingToneRemoteVolume>50</recordingToneRemoteVolume>
  <recordingToneDuration></recordingToneDuration>
  <displayOnWhenIncomingCall>0</displayOnWhenIncomingCall>
  <rtcp>0</rtcp>
  <moreKeyReversionTimer>5</moreKeyReversionTimer>
  <autoCallSelect>1</autoCallSelect>
  <g722CodecSupport>0</g722CodecSupport>
  <headsetWidebandUIControl>0</headsetWidebandUIControl>
  <handsetWidebandUIControl>0</handsetWidebandUIControl>
  <headsetWidebandEnable>0</headsetWidebandEnable>
  <handsetWidebandEnable>0</handsetWidebandEnable>
  <peerFirmwareSharing>0</peerFirmwareSharing>
  <enableCdpSwPort>0</enableCdpSwPort>
  <enableCdpPcPort>0</enableCdpPcPort>
  </vendorConfig>
  <userLocale>
  <name>Russian_Russia</name>
  <version>5.0(2)</version>
  <langCode>ru</langCode>
  <winCharSet>utf-8</winCharSet>
  </userLocale>
  <networkLocale>Russian_Federation</networkLocale>
  <transportLayerProtocol>2</transportLayerProtocol>

Входящие звонки на Cisco 7911 и Asterisk [закрыт]

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