Cisco G7962 + Astersk

BooM · Сообщение **BooM** » 09 авг 2011, 17:56

Днеь добрый, не как не хочет Cisco дружить с Asterisk'om.

Есть XMLDefault.cnf.xml со следующим содержанием:

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

<device>
    <fullConfig>true</fullConfig>
    <deviceProtocol>SIP</deviceProtocol>
    <devicePool>
        <dateTimeSetting>
            <dateTemplate>D.M.Y</dateTemplate>
            <timeZone>Ekaterinburg Standard Time</timeZone>
            <ntps>
                <ntp>
                    <name>192.168.88.13</name>
                    <ntpMode>Unicast</ntpMode>
                </ntp>
            </ntps>
        </dateTimeSetting>
        <callManagerGroup>
            <tftpDefault>true</tftpDefault>
                <members>
                <member priority="0">
                <callManager>
                <name>$ASTERISK</name>
                <description>CallManager 5.0</description>
                <ports>
                  <ethernetPhonePort>2000</ethernetPhonePort>
                  <sipPort>5060</sipPort>
                  <securedSipPort>5061</securedSipPort>
                </ports>
                <processNodeName>192.168.88.13</processNodeName>
                </callManager>
                </member>
                </members>
             </callManagerGroup>
    </devicePool>
    <commonProfile>
        <phonePassword></phonePassword>
        <backgroundImageAccess>true</backgroundImageAccess>
        <callLogBlfEnabled>0</callLogBlfEnabled>
    </commonProfile>
    <loadInformation>SIP42.8-5-4S</loadInformation>
    <loadInformation434  model="Cisco 7942">SIP42.8-5-4S</loadInformation434>
    <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>
        <daysDisplayNotActive>1,7</daysDisplayNotActive>
        <displayOnTime>10:30</displayOnTime>
        <displayOnDuration>06:05</displayOnDuration>
        <displayIdleTimeout>00:05</displayIdleTimeout>
        <webAccess>1</webAccess>
        <spanToPCPort>1</spanToPCPort>
        <loggingDisplay>1</loggingDisplay>
        <loadServer></loadServer>
    </vendorConfig>

<userLocale> 
  <name>Russian_Russian_Federation</name> 
  <uid></uid> 
  <langCode>ru_RU</langCode> 
  <version>8.4.3.1000-1</version> 
  <winCharSet>utf-8</winCharSet> 
</userLocale>


<networkLocale>Russian_Federation</networkLocale> 
<networkLocaleInfo> 
  <name>Russian_Federation</name> 
  <uid></uid> 
  <version>8.4.3.1000-1</version> 
</networkLocaleInfo>
    
    <deviceSecurityMode>1</deviceSecurityMode>
    <idleTimeout>0</idleTimeout>
    <directoryURL></directoryURL> 
    <servicesURL></servicesURL> 
    <idleURL></idleURL>
    <messagesURL></messagesURL>
    <proxyServerURL></proxyServerURL>
    <dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
    <dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
    <dscpForCm2Dvce>96</dscpForCm2Dvce>
    <transportLayerProtocol>2</transportLayerProtocol>
    <capfAuthMode>0</capfAuthMode>
    <capfList>
        <capf>
            <phonePort>3804</phonePort>
        </capf>
    </capfList>
    <certHash></certHash>
    <encrConfig>false</encrConfig>
    <sipProfile>
        <sipProxies>
            <backupProxy>192.168.88.13</backupProxy>
            <backupProxyPort>5060</backupProxyPort>
            <emergencyProxy>192.168.88.13</emergencyProxy>
            <emergencyProxyPort>5060</emergencyProxyPort>
            <outboundProxy>192.168.88.13</outboundProxy>
            <outboundProxyPort>5060</outboundProxyPort>
            <registerWithProxy>true</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> 
        <preferredCodec>g711alaw</preferredCodec>
       <dtmfAvtPayload>101</dtmfAvtPayload>
       <dtmfDbLevel>3</dtmfDbLevel>
       <dtmfOutofBand>avt</dtmfOutofBand>
        <alwaysUsePrimeLine>false</alwaysUsePrimeLine>
        <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
        <kpml>3</kpml>
        <stutterMsgWaiting>1</stutterMsgWaiting>
        <callStats>true</callStats>
        <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
        <disableLocalSpeedDialConfig>true</disableLocalSpeedDialConfig>
        <startMediaPort>10100</startMediaPort>
        <stopMediaPort>10300</stopMediaPort>
        <voipControlPort>5060</voipControlPort>
        <dscpForAudio>184</dscpForAudio>
        <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
        <dialTemplate>dialplan.xml</dialTemplate> 
        <phoneLabel>Cisco</phoneLabel>
          <natReceivedProcessing>false</natReceivedProcessing>
          <natEnabled>false</natEnabled>
          <natAddress></natAddress>
        <sipLines>
          <line button="1">
            <featureID>9</featureID>
            <featureLabel>700</featureLabel>
            <proxy>192.168.88.13</proxy>
            <port>5060</port>
            <name>700</name>
            <displayName>700</displayName>
            <autoAnswer>
              <autoAnswerEnabled>2</autoAnswerEnabled>
            </autoAnswer>
            <callWaiting>3</callWaiting>
            <authName>700</authName>
            <authPassword>cisco123</authPassword>
            <sharedLine>false</sharedLine>
            <messageWaitingLampPolicy>3</messageWaitingLampPolicy>
            <messagesNumber></messagesNumber>
            <ringSettingIdle>4</ringSettingIdle>
            <ringSettingActive>5</ringSettingActive>
            <contact>$ACCOUNT</contact>
            <forwardCallInfoDisplay>
              <callerName>true</callerName>
              <callerNumber>false</callerNumber>
              <redirectedNumber>false</redirectedNumber>
              <dialedNumber>true</dialedNumber>
            </forwardCallInfoDisplay>
          </line>
          <line button="2">
          <featureID></featureID>
          <featureLabel></featureLabel>
          <speedDialNumber></speedDialNumber>
          </line>
        </sipLines>
    </sipProfile>
</device>

Уже и локаль русская и прошивка SIP42.8-5-4S. Да вот только не приходят запросы о регистрации на Asterisk. Линия в телефоне появилась, но не активная. В ручную в аппарате тоже не могу ничего прописать. Разблокировку настроек делал.

Vlad1983 · Сообщение **Vlad1983** » 09 авг 2011, 18:09

для начала прошиться

как прошиваться и настраивать можно почитать здесь
http://bevice.ya.ru/replies.xml?item_no=131

ded · Сообщение **ded** » 09 авг 2011, 22:46

Vld1983 писал(а):для начала прошиться

У человека всё прошито уже. Boom - Надо создать SEP000123456789BCDE.cnf.xml. соответствующий МАС адресу телефона, положить в ТФТП directory и мониторить tcpdump port tftp чтобы видно было, что он её забирает.

Vlad1983 · Сообщение **Vlad1983** » 10 авг 2011, 07:06

от более свежей прошивки хуже обычно не бывает

BooM · Сообщение **BooM** » 10 авг 2011, 12:25

Прошился на новую прошивку. Скормил два файлика.

XMLDefault.cnf.xml

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

Код: Выделить всё

<Default>

<callManagerGroup>

<members>
      <member  priority="0">

<callManager>
          <ports>
            <ethernetPhonePort>2000</ethernetPhonePort>
            <mgcpPorts>

<listen>2427</listen>
              <keepAlive>2428</keepAlive>
            </mgcpPorts>

</ports>
          <processNodeName></processNodeName>
        </callManager>

</member>
    </members>

</callManagerGroup>


<authenticationURL></authenticationURL>
  <directoryURL></directoryURL>
  <idleURL></idleURL>

 <informationURL></informationURL>
  <messagesURL></messagesURL>
  <servicesURL></servicesURL>


</Default>

SEPE84040A2731C.cnf.xml

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

Код: Выделить всё

<device  xsi:type="axl:XIPPhone" ctiid="94">

<fullConfig>true</fullConfig>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>bevice</sshUserId>
<sshPassword></sshPassword>

<devicePool>
        <dateTimeSetting>
                <dateTemplate>YY-M-D</dateTemplate>
                <timeZone>Russian Standard/Daylight Time</timeZone>
        </dateTimeSetting>

        <callManagerGroup>
                <members> <member priority="0"> <callManager>
                        <name>192.168.88.13</name>
                        <ports>
                                <ethernetPhonePort>2000</ethernetPhonePort>
                                <sipPort>5060</sipPort>
                                <securedSipPort>5061</securedSipPort>
                        </ports>
                        <processNodeName>192.168.88.13</processNodeName>
                </callManager> </member> </members>
        </callManagerGroup>

        <srstInfo>
                <srstOption>Disable</srstOption>
                <ipAddr1></ipAddr1> <port1>2000</port1>
                <ipAddr2></ipAddr2> <port2>2000</port2>
                <ipAddr3></ipAddr3> <port3>2000</port3>
                <sipIpAddr1></sipIpAddr1> <sipPort1>5060</sipPort1>
                <sipIpAddr2></sipIpAddr2> <sipPort2>5060</sipPort2>
                <sipIpAddr3></sipIpAddr3> <sipPort3>5060</sipPort3>
                <isSecure>false</isSecure>
        </srstInfo>

        <connectionMonitorDuration>120</connectionMonitorDuration>
</devicePool>

<sipProfile>
        <sipProxies>
                <backupProxy></backupProxy> <backupProxyPort></backupProxyPort>
                <emergencyProxy></emergencyProxy> <emergencyProxyPort></emergencyProxyPort>
                <outboundProxy></outboundProxy> <outboundProxyPort></outboundProxyPort>
                <registerWithProxy>true</registerWithProxy>
        </sipProxies>
        <sipCallFeatures>
                <cnfJoinEnabled>true</cnfJoinEnabled>
                <callForwardURI>x-cisco-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>
                <retainForwardInformation>false</retainForwardInformation>
        </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>true</remotePartyID>
                <userInfo>None</userInfo>
        </sipStack>

        <autoAnswerTimer>1</autoAnswerTimer>
        <autoAnswerAltBehavior>false</autoAnswerAltBehavior>
        <autoAnswerOverride>true</autoAnswerOverride>
        <transferOnhookEnabled>false</transferOnhookEnabled>
        <enableVad>false</enableVad>
        <preferredCodec>none</preferredCodec>
        <dtmfAvtPayload>101</dtmfAvtPayload>
        <dtmfDbLevel>3</dtmfDbLevel>

        <dtmfOutofBand>avt</dtmfOutofBand>
        <kpml>3</kpml>
        <alwaysUsePrimeLine>false</alwaysUsePrimeLine>
        <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
        <phoneLabel></phoneLabel>
        <stutterMsgWaiting>2</stutterMsgWaiting>
        <callStats>true</callStats>
        <offhookToFirstDigitTimer>15000</offhookToFirstDigitTimer>
        <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
        <disableLocalSpeedDialConfig>true</disableLocalSpeedDialConfig>
        <poundEndOfDial>false</poundEndOfDial>
        <startMediaPort>16384</startMediaPort>
        <stopMediaPort>32766</stopMediaPort>

        <sipLines>
                <line  button="1" lineIndex="1">
                        <featureID>9</featureID>
                        <proxy>192.168.88.13</proxy>
                        <port>5060</port>
                        <autoAnswer> <autoAnswerEnabled>2</autoAnswerEnabled> </autoAnswer>
                        <callWaiting>3</callWaiting>

                        <sharedLine>false</sharedLine>
                        <messageWaitingLampPolicy>3</messageWaitingLampPolicy>
                        <messagesNumber></messagesNumber>
                        <ringSettingIdle>4</ringSettingIdle>
                        <ringSettingActive>5</ringSettingActive>
                        <forwardCallInfoDisplay>
                                <callerName>true</callerName>
                                <callerNumber>false</callerNumber>
                                <redirectedNumber>false</redirectedNumber>
                                <dialedNumber>true</dialedNumber>
                        </forwardCallInfoDisplay>

                        <featureLabel>700</featureLabel>
                        <displayName>700</displayName>
                        <name>700</name>
                        <authName>700</authName>
                        <authPassword></authPassword>
                </line>
        </sipLines>
        <externalNumberMask>700</externalNumberMask>


        <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>SIP42.9-2-1S</loadInformation>

<vendorConfig>
        <disableSpeaker>false</disableSpeaker>
        <disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
        <forwardingDelay>1</forwardingDelay>
        <pcPort>0</pcPort>
        <settingsAccess>1</settingsAccess>
        <garp>0</garp>
        <voiceVlanAccess>0</voiceVlanAccess>
        <videoCapability>0</videoCapability>
        <autoSelectLineEnable>1</autoSelectLineEnable>
        <webAccess>1</webAccess>
        <daysDisplayNotActive>1,7</daysDisplayNotActive>
        <displayOnTime>09:00</displayOnTime>
        <displayOnDuration>12:00</displayOnDuration>
        <displayIdleTimeout>01:00</displayIdleTimeout>
        <spanToPCPort>1</spanToPCPort>
        <loggingDisplay>2</loggingDisplay>
        <loadServer>192.168.88.13</loadServer>
        <recordingTone>0</recordingTone>
        <recordingToneLocalVolume>100</recordingToneLocalVolume>
        <recordingToneRemoteVolume>50</recordingToneRemoteVolume>
        <recordingToneDuration></recordingToneDuration>
        <displayOnWhenIncomingCall>0</displayOnWhenIncomingCall>
        <rtcp>0</rtcp>
        <moreKeyReversionTimer>5</moreKeyReversionTimer>
        <autoCallSelect>1</autoCallSelect>
        <logServer>192.168.88.13</logServer>
        <g722CodecSupport>1</g722CodecSupport>
        <headsetWidebandUIControl>1</headsetWidebandUIControl>
        <handsetWidebandUIControl>1</handsetWidebandUIControl>
        <headsetWidebandEnable>1</headsetWidebandEnable>
        <handsetWidebandEnable>1</handsetWidebandEnable>
        <peerFirmwareSharing>0</peerFirmwareSharing>
        <enableCdpSwPort>1</enableCdpSwPort>
        <enableCdpPcPort>1</enableCdpPcPort>
</vendorConfig>

<versionStamp> 2010-01-24 22:42</versionStamp>

<userLocale> 
  <name>Russian_Russian_Federation</name> 
  <uid></uid> 
  <langCode>ru_RU</langCode> 
  <version>8.4.3.1000-1</version> 
  <winCharSet>utf-8</winCharSet> 
</userLocale>


<networkLocale>Russian_Federation</networkLocale> 
<networkLocaleInfo> 
  <name>Russian_Federation</name> 
  <uid></uid> 
  <version>8.4.3.1000-1</version> 
</networkLocaleInfo>

<deviceSecurityMode>1</deviceSecurityMode>
<idleTimeout>0</idleTimeout>
<authenticationURL>http://192.168.88.13/auth.php</authenticationURL>
<directoryURL>http://192.168.88.13/dirs.php</directoryURL>
<idleURL></idleURL>
<informationURL>http://192.168.88.13/info.php</informationURL>
<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<servicesURL>http://192.168.88.13/service.php</servicesURL>
<dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
<dscpForCm2Dvce>96</dscpForCm2Dvce>
<transportLayerProtocol>4</transportLayerProtocol>
<singleButtonBarge>0</singleButtonBarge>

<capfAuthMode>0</capfAuthMode>
<capfList> <capf>
        <phonePort>3804</phonePort>
        <processNodeName>192.168.88.13</processNodeName>
</capf> </capfList>

<certHash></certHash>
<encrConfig>false</encrConfig>
<advertiseG722Codec>1</advertiseG722Codec>

</device>

Файлы закачиваются, эффект тот же, линия создаётся, горит "регистрация", не каких инвайтов и запросов об авторизации в Asterisk не приходит.

ded · Сообщение **ded** » 10 авг 2011, 12:40

Влом проверять Ваш конфиг файл, не увидел параметров линий там, наверняка ошибки, поэтому не регистрируется. This is the sample -
http://asterisk.ru/knowledgebase/cisco7970
Надо залазить в телефон по SSH и смотреть логи, можно конечно смотреть через Status messages on display, но это менее информативно.

Vlad1983 · Сообщение **Vlad1983** » 10 авг 2011, 12:47

в SEPE84040A2731C.cnf.xml
нет первой строки

Код: Выделить всё

<?xml version="1.0" encoding="UTF-8"?>

у мну стояло так

Код: Выделить всё

...
                      <featureLabel>   </featureLabel>
...
        <externalNumberMask></externalNumberMask>
...

убрать из всех файлов

Код: Выделить всё

<userLocale>
<name>Russian_Russian_Federation</name>
<langCode>ru_RU</langCode>
<version></version>
<winCharSet>utf-8</winCharSet>
</userLocale>

<networkLocale>Russian_Federation</networkLocale>
<networkLocaleInfo>
<name>Russian_Federation</name>
<uid></uid>
<version></version>
</networkLocaleInfo>

и смотрим на серваке не в консоль астериска, а снифер
tcpdump -i any -vnn -s0 port 5060
чую ты там увидишь что будет ломиться на порт 5060, но на TCP, а у тя астер слушает 5060 UDP

короче настроить астер проверить регистрацию и звонки.
по том уже всякие эксперименты с локалью и т.д.

BooM · Сообщение **BooM** » 10 авг 2011, 17:09

Залил конфиг по ссылке http://asterisk.ru/knowledgebase/cisco7970, вставил свои параметры.

SEPE84040A2731C.cnf.xml

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

Код: Выделить всё

<?xml version="1.0" encoding="UTF-8"?>
<device> 
<deviceProtocol>SIP</deviceProtocol> 
<sshUserId>admin</sshUserId> 
<sshPassword>12345</sshPassword> 
<devicePool> 
   <dateTimeSetting> 
      <dateTemplate>D/M/YA</dateTemplate> 
      <timeZone>Central Standard/Daylight Time</timeZone> 
      <ntps> 
         <ntp> 
            <name>europe.pool.ntp.org</name> 
            <ntpMode>Unicast</ntpMode> 
         </ntp>         
      </ntps> 
   </dateTimeSetting> 
   <callManagerGroup> 
      <members> 
         <member priority="0"> 
            <callManager> 
               <ports> 
                  <ethernetPhonePort>2000</ethernetPhonePort> 
                  <sipPort>5060</sipPort> 
                  <securedSipPort>5061</securedSipPort> 
               </ports> 
               <processNodeName>192.168.88.13</processNodeName> 
            </callManager> 
         </member> 
      </members> 
   </callManagerGroup> 
</devicePool> 
<sipProfile> 
   <sipProxies> 
      <backupProxy></backupProxy> 
      <backupProxyPort>5060</backupProxyPort> 
      <emergencyProxy></emergencyProxy> 
      <emergencyProxyPort></emergencyProxyPort> 
      <outboundProxy></outboundProxy> 
      <outboundProxyPort></outboundProxyPort> 
      <registerWithProxy>true</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>PHONE TITLE</phoneLabel> 
   <stutterMsgWaiting>1</stutterMsgWaiting> 
   <callStats>false</callStats> 
   <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts> 
   <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig> 
   <sipLines> 
      <line button="1"> 
         <featureID>9</featureID> 
         <featureLabel>700</featureLabel> 
         <proxy>192.168.88.13</proxy> 
         <port>5060</port> 
         <name>700</name> 
         <displayName>700</displayName> 
         <autoAnswer> 
            <autoAnswerEnabled>2</autoAnswerEnabled> 
         </autoAnswer> 
         <callWaiting>3</callWaiting> 
         <authName>700</authName> 
         <authPassword>cisco123</authPassword> 
         <sharedLine>false</sharedLine> 
         <messageWaitingLampPolicy>1</messageWaitingLampPolicy> 
         <messagesNumber>*99</messagesNumber> 
         <ringSettingIdle>4</ringSettingIdle> 
         <ringSettingActive>5</ringSettingActive> 
         <contact>700</contact> 
         <forwardCallInfoDisplay> 
            <callerName>true</callerName> 
            <callerNumber>false</callerNumber> 
            <redirectedNumber>false</redirectedNumber> 
            <dialedNumber>true</dialedNumber> 
         </forwardCallInfoDisplay> 
      </line> 
        </sipLines>
 
   <voipControlPort>5060</voipControlPort> 
   <startMediaPort>16348</startMediaPort> 
   <stopMediaPort>20134</stopMediaPort> 
   <dscpForAudio>184</dscpForAudio> 
   <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy> 
   <dialTemplate>dialplan.xml</dialTemplate> 
   <softKeyFile></softKeyFile> 
</sipProfile> 
<commonProfile> 
   <phonePassword></phonePassword> 
   <backgroundImageAccess>true</backgroundImageAccess> 
   <callLogBlfEnabled>2</callLogBlfEnabled> 
</commonProfile> 
<loadInformation>SIP42.9-2-1S</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> 
<userLocale> 
   <name></name> 
   <uid></uid> 
   <langCode>en_US</langCode> 
   <version>1.0.0.0-1</version> 
   <winCharSet>iso-8859-1</winCharSet> 
</userLocale> 
<networkLocale></networkLocale> 
<networkLocaleInfo> 
   <name></name> 
   <uid></uid> 
   <version>1.0.0.0-1</version> 
</networkLocaleInfo>    
<deviceSecurityMode>1</deviceSecurityMode> 
<authenticationURL></authenticationURL> 
<directoryURL></directoryURL> 
<servicesURL></servicesURL> 
<idleURL></idleURL> 
<informationURL></informationURL> 
<messagesURL></messagesURL> 
<proxyServerURL></proxyServerURL> 
<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>

В дампе с IP'шником телефона только всевозможные Options

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

16:57:54.942047 IP (tos 0x60, ttl 64, id 39074, offset 0, flags [none], proto: UDP (17), length: 540) 192.168.88.13.5060 > 192.168.88.17.5060: SIP, length: 512
OPTIONS sip:192.168.88.17 SIP/2.0
Via: SIP/2.0/UDP 192.168.88.13:5060;branch=z9hG4bK5a36e6c4;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.88.13>;tag=as7bcd4b18
To: <sip:192.168.88.17>
Contact: <sip:Unknown@192.168.88.13>
Call-ID: 2a82ac5753bb38ae63ca3c8d02508bc4@192.168.88.13
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.6.2.13)
Date: Wed, 10 Aug 2011 12:57:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

Попробовал на SSH постучаться через Putty, ничего не выходит. В статусе телефона единственная ошибка Error Updating Locale, но это из за того что путь к локали не прописывал. Phone Still Registering...

ЗЫ
Ещё в Sip debug выловил такое

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

<--- SIP read from UDP:192.168.88.17:50146 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.88.13:5060;branch=z9hG4bK2280e9c0;rport
From: "Unknown" <sip:Unknown@192.168.88.13>;tag=as0758190f
To: <sip:192.168.88.17>;tag=e84040a2731c0005758ceb17-4a92c001
Call-ID: 5ff8acbd47ba70545ad9795c57003a9f@192.168.88.13
Date: Wed, 11 May 2011 20:13:38 GMT
CSeq: 102 OPTIONS
Server: Cisco-CP7962G/9.2.1
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Allow-Events: kpml,dialog,refer
Accept: application/sdp,multipart/mixed,multipart/alternative
Accept-Encoding: identity
Accept-Language: en
Supported: replaces,join,sdp-anat,norefersub
Content-Length: 314
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 23590 0 IN IP4 192.168.88.17
s=SIP Call
t=0 0
m=audio 0 RTP/AVP 0 8 18 102 116 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 L16/16000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

Выходит всё таки по UDP пакеты бегают... только вот почему инвайты не высылаются, загадка... Я вот думаю, а в портах не каких не может быть дело, Cisco какие то дополнительные порты нада открыть или нет?

ded · Сообщение **ded** » 10 авг 2011, 17:22

Это Вы выловили пакеты OPTIONS - не влияют на ситуацию.
Я вспомнил проблему с супер новыми моделями Cisco phones: надо в Астериске включить ТСР и он зарегистрируется по ТСР.

BooM · Сообщение **BooM** » 10 авг 2011, 17:42

Прописал в sip_custom.conf

Код: Выделить всё

tcpenable=yes
transport=tcp

Options теперь гоняются по TCP протоколу с приставочкой transport=tcp.
Регистрацию так и не проходит.

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