[Gelöst] 7961 an externem VoIP Server ( SIP Image )

dynamic · 22 Jun 2012

Hallo,
ich bekomme mein 7961 ( SIP Image, kein Callmanager ) einfach nicht dazu sich an einem externen Asterisk Server zu registrieren.

An dem gleichen Asterisk-Server kann ich mich z.B. mittels CSipSimple von meinem Android-Phone aus ( sowohl aus gleichem LAN / WLAN oder via Internet ) problemlos registrieren. Daher vermute ich ein Problem im angehängten Sepxxxxx.cnf.xml Config-File.

Code:

<device>
    <deviceProtocol>SIP</deviceProtocol>
    <sshUserId>dynamic</sshUserId>
    <sshPassword>abcdef1234</sshPassword>
    <devicePool>
        <dateTimeSetting>
            <name>CMLocal</name>
            <dateTemplate>Y/M/D</dateTemplate>
            <timeZone>W. Europe Standard/Daylight Time</timeZone>
            <ntps>
                <ntp>
                    <name>2.de.pool.ntp.org</name>
                    <ntpMode>Unicast</ntpMode>
                </ntp>
                <ntp>
                    <name>131.234.137.23</name> 
                    <ntpMode>Unicast</ntpMode> 
                
                </ntp> 
                <ntp>
                      <name>192.53.103.108</name> 
                      <ntpMode>Unicast</ntpMode> 
                </ntp>
            </ntps>
        </dateTimeSetting>
        <callManagerGroup>
            <members>
                <member>
                    <callManager>
                        <ports>
                            <ethernetPhonePort>2000</ethernetPhonePort>
                            <sipPort>5060</sipPort>
                            <securedSipPort>5060</securedSipPort> 
                        </ports>
                        <processNodeName>xxxxx.homeip.net</processNodeName>
                    </callManager>
                </member>
            </members>
        </callManagerGroup>
    </devicePool>
    <sipProfile>
        <preferredCodec>g711alaw</preferredCodec>
        <sipProxies>
            <backupProxy></backupProxy>
            <backupProxyPort></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>60</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>
        <dtmfAvtPayload>101</dtmfAvtPayload>
        <dtmfDbLevel>3</dtmfDbLevel>
        <dtmfOutofBand>avt</dtmfOutofBand>
        <alwaysUsePrimeLine>false</alwaysUsePrimeLine>
        <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
        <kpml>3</kpml>
        <natReceivedProcessing>false</natReceivedProcessing>
        <natEnabled>false</natEnabled>
        <natAddress>demirel.homeip.net</natAddress> 
        <phoneLabel>Home Office</phoneLabel>
        <stutterMsgWaiting>1</stutterMsgWaiting>
        <callStats>false</callStats>
        <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
        <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
        <startMediaPort>16384</startMediaPort>
        <stopMediaPort>32766</stopMediaPort>
        <sipLines>
            <line button="1">
                <featureID>9</featureID>
                <featureLabel>+49xxxxxxxxx</featureLabel>
                <proxy>xxxxx.homeip.net</proxy>
                <port>5062</port>
                <name>779</name>
                <displayName>779</displayName>
                <autoAnswer>
                    <autoAnswerEnabled>2</autoAnswerEnabled>
                </autoAnswer>
                <callWaiting>3</callWaiting>
                <authName>779</authName>
                <authPassword>779</authPassword>
                <sharedLine>true</sharedLine>
                <messageWaitingLampPolicy>1</messageWaitingLampPolicy>
                <messagesNumber>9779</messagesNumber>
                <ringSettingIdle>4</ringSettingIdle>
                <ringSettingActive>5</ringSettingActive>
                <contact>779</contact>
                <forwardCallInfoDisplay>
                    <callerName>false</callerName>
                    <callerNumber>true</callerNumber>
                    <redirectedNumber>true</redirectedNumber>
                    <dialedNumber>true</dialedNumber>
                </forwardCallInfoDisplay>
            </line>    
            <line button="4">
                <featureID>21</featureID>
                <featureLabel>Name</featureLabel>
                <speedDialNumber>061xxxxxx</speedDialNumber>
            </line>
            <line button="5">
                <featureID>21</featureID>
                <featureLabel>Conf DE</featureLabel>
                <speedDialNumber>069xxxxx</speedDialNumber>
            </line>
            <line button="6">
                <featureID>21</featureID>
                <featureLabel>Conf UK</featureLabel>
                <speedDialNumber>0044xxxxxxx</speedDialNumber>
            </line>
        </sipLines>
        <voipControlPort>5060</voipControlPort>
        <dscpForAudio>184</dscpForAudio>
        <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
        <dialTemplate>DRdialplan.xml</dialTemplate>
        <softKeyFile>softkey.xml</softKeyFile> 
    </sipProfile>
    <commonProfile>
        <phonePassword>71</phonePassword>
        <backgroundImageAccess>TRUE</backgroundImageAccess>
        <callLogBlfEnabled>2</callLogBlfEnabled>
    </commonProfile>
    <loadInformation>SIP41.8-5-4S</loadInformation>
    <vendorConfig>
        <daysDisplayNotActive>2,3,4,5,6,7</daysDisplayNotActive>
        <displayOnTime>08:30</displayOnTime>
        <displayOnDuration>11:30</displayOnDuration>
        <displayIdleTimeout>00:05</displayIdleTimeout>
        <spanToPCPort>0</spanToPCPort>
        <loggingDisplay>1</loggingDisplay>
        <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>0</webAccess>
    </vendorConfig>
    <networkLocale>Germany</networkLocale> 
    <networkLocaleInfo> 
        <name>Germany</name> 
        <version>x.x(x)</version> 
        <uid></uid> 
    </networkLocaleInfo> 
    <deviceSecurityMode>0</deviceSecurityMode>
    <authenticationURL>http://www.xxxxxxx.xxx/auth.php</authenticationURL>
    <directoryURL></directoryURL>
    <idleURL>http://www.xxxxxxx.xxx/xxx.php</idleURL>
    <informationURL></informationURL> 
    <messagesURL></messagesURL>
    <servicesURL>http://www.xxxxxxx.xxx/cisco_service_index.php</servicesURL>
    <proxyServerURL></proxyServerURL>
</device>

Die Logs zeigen, daß die Authentifiziering im 3. Schritt unterbrochen wird ( siehe schematische Darstellung ), da das 7961 nicht erwartungsgemäß mit dem Register + Authroization String ( d.h. ID und hashed PW ) antwortet.

Das SIP debug-log für OK und NOK Fall hänge ich hier ebenfalls bei ...

Code:

<--- SIP read from UDP:78.35.235.191:51661 --->
REGISTER sip:xxxxxx.homeip.net SIP/2.0
Via: SIP/2.0/UDP 192.168.178.199:5060;branch=z9hG4bKf05ddff3
From: <sip:[email protected]>;tag=00170e617de72c4cd679ecc7-ff086743
To: <sip:[email protected]>
Call-ID: [email protected]
Max-Forwards: 70
Date: Fri, 22 Jun 2012 14:19:48 GMT
CSeq: 5273 REGISTER
User-Agent: Cisco-CP7961G/8.5.3
Contact: <sip:[email protected]:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-00170e617de7>";+u.sip!model.ccm.cisco.com="30018"
Supported: (null),X-cisco-xsi-7.0.1
Content-Length: 0
Expires: 60

<------------->
--- (13 headers 0 lines) ---
Sending to 78.35.235.191:5060 (no NAT)

<--- Transmitting (no NAT) to 78.35.235.191:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.178.199:5060;branch=z9hG4bKf05ddff3;received=78.35.235.191
From: <sip:[email protected]>;tag=00170e617de72c4cd679ecc7-ff086743
To: <sip:[email protected]>;tag=as52d5abd1
Call-ID: [email protected]
CSeq: 5273 REGISTER
Server: Asterisk PBX 10.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2fd4799e"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:78.35.235.191:51661 --->
REGISTER sip:xxxxxx.homeip.net SIP/2.0
Via: SIP/2.0/UDP 192.168.178.199:5060;branch=z9hG4bKf05ddff3
From: <sip:[email protected]>;tag=00170e617de72c4cd679ecc7-ff086743
To: <sip:[email protected]>
Call-ID: [email protected]
Max-Forwards: 70
Date: Fri, 22 Jun 2012 14:19:48 GMT
CSeq: 5273 REGISTER
User-Agent: Cisco-CP7961G/8.5.3
Contact: <sip:[email protected]:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-00170e617de7>";+u.sip!model.ccm.cisco.com="30018"
Supported: (null),X-cisco-xsi-7.0.1
Content-Length: 0
Expires: 60
[COLOR=#ff0000]
############ WIE ES BEI CSIPSIMPLE AUSSIEHT #################[/COLOR]

<--- SIP read from UDP:xx.xx.235.191:43721 --->
REGISTER sip:xxxxxx.homeip.net:5062 SIP/2.0
Via: SIP/2.0/UDP xx.xx.235.191:43721;rport;branch=z9hG4bKPjgwpZJBtPGemXhTPgpxW4fPU3ngFhjwbb
Route: <sip:xxxxxx.homeip.net:5062;transport=udp;lr>
Max-Forwards: 70
From: <sip:[email protected]>;tag=2I4T8qHn14qHDFOX4YAhTVfm6QnjUgja
To: <sip:[email protected]>
Call-ID: N2kIMe-YI.SmUboLMPM4NwmURHctccgT
CSeq: 48514 REGISTER
User-Agent: CSipSimple r1108 / GT-I9100-15
Contact: <sip:[email protected]:43721;ob>
Expires: 900
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to xx.xx.235.191:43721 (no NAT)

<--- Transmitting (no NAT) to xx.xx.235.191:43721 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP xx.xx.235.191:43721;branch=z9hG4bKPjgwpZJBtPGemXhTPgpxW4fPU3ngFhjwbb;received=xx.xx.235.191;rport=43721
From: <sip:[email protected]>;tag=2I4T8qHn14qHDFOX4YAhTVfm6QnjUgja
To: <sip:[email protected]>;tag=as0f2dec51
Call-ID: N2kIMe-YI.SmUboLMPM4NwmURHctccgT
CSeq: 48514 REGISTER
Server: Asterisk PBX 10.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2992f011"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'N2kIMe-YI.SmUboLMPM4NwmURHctccgT' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:xx.xx.235.191:43721 --->
REGISTER sip:xxxxxx.homeip.net:5062 SIP/2.0
Via: SIP/2.0/UDP xx.xx.235.191:43721;rport;branch=z9hG4bKPj.zBbuMUc7XUHFrtAmKfqC92xD5guOlFc
Route: <sip:xxxxxx.homeip.net:5062;transport=udp;lr>
Max-Forwards: 70
From: <sip:[email protected]>;tag=2I4T8qHn14qHDFOX4YAhTVfm6QnjUgja
To: <sip:[email protected]>
Call-ID: N2kIMe-YI.SmUboLMPM4NwmURHctccgT
CSeq: 48515 REGISTER
User-Agent: CSipSimple r1108 / GT-I9100-15
Contact: <sip:[email protected]:43721;ob>
Expires: 900
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
[B][COLOR=#ff0000]Authorization: Digest username="779", realm="asterisk", nonce="2992f011", uri="sip:xxxxxx.homeip.net:5062", response="e803b048f36d08f472245d469562f7a3", algorithm=MD5
Content-Length: 0[/COLOR]
[/B]
<------------->
--- (14 headers 0 lines) ---
Sending to xx.xx.235.191:43721 (no NAT)
    -- Registered SIP '779' at xx.xx.235.191:43721
Reliably Transmitting (no NAT) to xx.xx.235.191:43721:
OPTIONS sip:[email protected]:43721;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.178.64:5062;branch=z9hG4bK6d411c39
Max-Forwards: 70
From: "asterisk" <sip:[email protected]:5062>;tag=as41551521
To: <sip:[email protected]:43721;ob>
Contact: <sip:[email protected]:5062>
Call-ID: [email protected]:5062
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.3.0
Date: Fri, 22 Jun 2012 11:19:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
       > Saved useragent "CSipSimple r1108 / GT-I9100-15" for peer 779

<--- Transmitting (no NAT) to xx.xx.235.191:43721 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.235.191:43721;branch=z9hG4bKPj.zBbuMUc7XUHFrtAmKfqC92xD5guOlFc;received=xx.xx.235.191;rport=43721
From: <sip:[email protected]>;tag=2I4T8qHn14qHDFOX4YAhTVfm6QnjUgja
To: <sip:[email protected]>;tag=as0f2dec51
Call-ID: N2kIMe-YI.SmUboLMPM4NwmURHctccgT
CSeq: 48515 REGISTER
Server: Asterisk PBX 10.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 900
Contact: <sip:[email protected]:43721;ob>;expires=900
Date: Fri, 22 Jun 2012 11:19:32 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'N2kIMe-YI.SmUboLMPM4NwmURHctccgT' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:xx.xx.235.191:43721 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.178.64:5062;received=xx.xx.235.191;branch=z9hG4bK6d411c39
Call-ID: [email protected]:5062
From: "asterisk" <sip:[email protected]>;tag=as41551521
To: <sip:[email protected];ob>;tag=z9hG4bK6d411c39
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
User-Agent: CSipSimple r1108 / GT-I9100-15
Content-Type: application/sdp
Content-Length: 383

v=0
o=- 3549352772 3549352772 IN IP4 192.168.178.55
s=pjmedia
c=IN IP4 192.168.178.55
t=0 0
m=audio 4000 RTP/AVP 9 107 106 105 0 8 101
a=rtcp:4001 IN IP4 192.168.178.55
a=rtpmap:9 G722/8000
a=rtpmap:107 speex/32000
a=rtpmap:106 speex/16000
a=rtpmap:105 speex/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (13 headers 16 lines) ---

Hat da jemand evtl. eine Idee ... irgendein Setting muss es sein !

Evtl. würde eine funktionierende SEPxxxxx.cnf.xml mit der sich ein 79x1 an einem externen VoIP Provider erfolgreich anmelden kann auch weiterhelfen ?
Das würde ggf. auch helfen.

Gruß
dynamic

dynamic · 24 Jun 2012

Möglicher Lösungsansatz

Wie es aussieht kann das Problem darauf zurückgeführt werden daß Cisco mit symmetrischem NAT nicht klar kommt bzw. diesen aus Sicherheitsgründen nicht verwendet.

Beispiel:
Cisco schickt den SIP Register Request von einem hohen Port ( > 49xxx ) erwartet aber die Antwort dennoch ausschließlich am konfigurierten <voipcontrolport>, was aber nur dann der Fall ist, wenn auf VoIP-Server ( Asterisk-Server ) Seite kein NAT genutzt wird.

Nach einigen Tests habe ich jetzt folgende Konfiguration, die funktionert ( hier führe ich lediglich die wichtigsten Parameter auf:

SEPxxxxx.cnf.xml:

Code:

...
<outboundProxyPort></outboundProxyPort>
<registerWithProxy>true</registerWithProxy>
...
<natReceivedProcessing>true</natReceivedProcessing>
<natEnabled>true</natEnabled>
<natAddress>[COLOR=#0000cd]#external.hostname.phone#[/COLOR]</natAddress> [COLOR=#ff0000]#externer Router-Hostname ( z.B. dydns-Adresse ) an dem das Telefon hängt [/COLOR]
...
<sipLines>
<proxy>[COLOR=#0000cd]#external.hostname.VoIPServer#[/COLOR]</proxy>
<port>[COLOR=#0000cd]#Server-VoIP-Port#[/COLOR]</port>
...
 <voipControlPort>[COLOR=#0000cd]#Voip-Control-Port-Phone#[/COLOR]</voipControlPort> [COLOR=#ff0000]#darauf achten, daß dies z.B. nicht Port 5060 ist, da dieser Port ( z.B. FritzBox ) bereits genutzt wird[/COLOR]
...

Router-Port-Forwardings ( VoIP-Client ):

Code:

[COLOR=#0000CD]#Voip-Control-Port-Phone# [/COLOR]an lokale IP des Cisco-Telefons

Router-Port-Forwardings ( VoIP-Server ):

Code:

[COLOR=#0000cd]#Server-VoIP-Port#[/COLOR] an lokale IP des VoIP/Asterisk-Servers

Asterisk-Settings ( sip.conf ):

Code:

nat=[COLOR=#0000cd]no[/COLOR]
externhost=[COLOR=#0000cd]#external.hostname.VoIPServer#[/COLOR]
localnet=[COLOR=#0000cd]192.168.178.0/255.255.255.0[/COLOR]

Ausgiebig getestet ist diese Konfiguration jedoch noch nicht ... evtl. werden hier noch weitere Updates erforderlich ( auch wenn es erstmal zu funktionieren scheint ).

Was mich noch stutzig macht sinde folgende zwei Meldungen auf Asterisk-Server Seite, die evtl. noch auf Problem hinweisen könnten:

Code:

Really destroying SIP dialog '3e8fd8ec5017cd4a6b0fff5a481c9252@[COLOR=#0000cd]#external.hostname.VoIPServer#[/COLOR]:[COLOR=#0000cd]#Server-VoIP-Port#[/COLOR]' Method: OPTIONS
Really destroying SIP dialog '00170e61-7de70003-dfcb9a1f-5e391894@[COLOR=#0000cd]#external.hostname.phone#[/COLOR]' Method: REGISTER

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[Gelöst] 7961 an externem VoIP Server ( SIP Image )

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