Need help with cisco 8961

csmathers
Posts: 9
Member Since:
2006-09-05

I am having issues with a new Cisco 8961 phones I am trying out. I have loaded the most recent firmware on the phone. it is 9-2-3-27. I have a working SEP file but I can't make incoming or outgoing phone calls. I get a dial tone but when I try to dial a number nothing happens. It does not matter if it is a local extension or outside number. I have SSH into the trixbox machine and watch the asterisk -r prompt but I don't see any logs when I try the call. Inside of trixbox on config file editor I have edited the sip_custom file to add these lines:
tcpenable=yes
transport=tcp,udp

For the extension I have the follwing settings...

This device uses sip technology.
secret=pass
dtmfmode=rfc2833
canreinvite =yes
context=from-internal
host=dynamic
type=friend
nat=never
port=5060
qualify=no
callgroug=
pickupgroup=
disallow=
allow=
dial=SIP/108
accountcode
mailbox=108@device
deny=0.0.0.0/0.0.0.0
permit=0.0.0.0/0.0.0.0

The phone Status Message screen shows.....

7:00:59p DNS Timeout
7:01:00p DNS Unknown Host example.domain.ext
7:01:00p Updating Trust List
7:01:00p No Trust List installed
7:01:01p SEPE80462EB0E37.cnf.xml (TFTP)
7:01:02p VPN Error: VPN is not Configured.
7:01:25p File Not Found : /gh-sip.jar
7:01:25p Error Updating Locale
7:01:25p File Not Found : /g4-tones.xml
7:01:25p Error Updating Locale


The xml config is....
<device> 
<deviceProtocol>SIP</deviceProtocol> 
<sshUserId>admin</sshUserId> 
<sshPassword>password</sshPassword> 
<devicePool> 
   <dateTimeSetting> 
      <dateTemplate>D/M/YA</dateTemplate> 
      <timeZone>Eastern Standard/Daylight Time</timeZone> 
      <ntps> 
         <ntp> 
            <name>192.168.10.4</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>192.168.10.4</processNodeName> 
            </callManager> 
         </member> 
      </members> 
   </callManagerGroup> 
</devicePool> 
<sipProfile>
   <natEnabled>false</natEnabled>
   <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>Home</phoneLabel> 
   <stutterMsgWaiting>1</stutterMsgWaiting> 
   <callStats>false</callStats> 
   <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts> 
   <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig> 
   <sipLines> 
      <line button="1"> 
         <featureID>9</featureID> 
         <featureLabel>108</featureLabel> 
         <proxy>192.168.10.4</proxy> 
         <port>5060</port> 
         <name>108</name> 
         <displayName>108</displayName> 
         <autoAnswer> 
            <autoAnswerEnabled>2</autoAnswerEnabled> 
         </autoAnswer> 
         <callWaiting>3</callWaiting> 
         <authName>108</authName> 
         <authPassword>pass</authPassword> 
         <sharedLine>false</sharedLine> 
         <messageWaitingLampPolicy>1</messageWaitingLampPolicy> 
         <messagesNumber>*99</messagesNumber> 
         <ringSettingIdle>4</ringSettingIdle> 
         <ringSettingActive>5</ringSettingActive> 
         <contact>108</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>sip8961.9-2-3-27</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>0</webAccess>
   <sshAccess>0</sshAccess>
   <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>http://example.domain.ext/services/authenticate.php</authenticationURL> 
<directoryURL>http://example.domain.ext/services/directory.php</directoryURL> 
<servicesURL>http://example.domain.ext/services/menu.xml</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>

I have a feeling the problem is with NAT. I have added false under the sipProfile but still nothing. Please help. I am at the end of my rope.

thanks



SkykingOH
Posts: 9561
Member Since:
2007-12-17
I fixed your XML code so it

I fixed your XML code so it displays properly, suggest you read the thread "how to ask for help" it contains all sots of tips on formatting and pulling in logs.

As far as your actual problem I am not familiar with that phone. I know that Asterisk 1.8 added features to support MWI on them.

--

Scott

aka "Skyking"



csmathers
Posts: 9
Member Since:
2006-09-05
Thank you

SkykingOH, thanks for fixing the xml!!! I am currently running Trixbox 2.8 with Asterisk 1.6. Is there a way to update asterisk engine to 1.8. I was able to run some debugging and notice I was getting a 401 UnAuthorized error when i tried to dial another extension. I have checked and rechecked the passwords and everything matches up. This is the log I get...

Retransmitting #6 (no NAT) to 10.0.1.103:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.1.103:5060;branch=z9hG4bK756fea75;received=10.0.1.103
From: "102" ;tag=e80462eb0e3700057261f071-5c31a0a2
To: ;tag=as66998fa3
Call-ID: e80462eb-0e370005-7f0e14b9-70013275@10.0.1.103
CSeq: 101 INVITE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2ced33a6"
Content-Length: 0

I tried called Trixbox support today but was on hold for 2 hours and then it went to voice mail.

Do you think I should just abandon trixbox and use Asterisk???



SkykingOH
Posts: 9561
Member Since:
2007-12-17
trixbox has not been updated

trixbox has not been updated or supported in years. What you called was the Digium support line for their commercial products.

Anyway, straight Asterisk would seem to be a bit overkill. trixbox is based on FreePBX and Asterisk. FreePBX team has a distro and PBX in a Flash is also current. Both offer Asterisk 1.8 support and the current version of FreePBX.

--

Scott

aka "Skyking"



csmathers
Posts: 9
Member Since:
2006-09-05
Update....

Well i have installed Freepbx on the server now but still same problems. I was able to downgrade the firmware and when i did that I was able to call another soft phone extension. However the phone was still not registering with server so the softphone could not call back. I used tcpdump src 10.0.1.103 -vvv as the phone booted up and this was the only thing that came up a couple of times....

10:16:50.840603 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto: UDP (17), length: 76) 10.0.1.103.ntp > 10.0.1.253.ntp: [udp sum ok] NTPv3, length 48
Client, Leap indicator: clock unsynchronized (192), Stratum 12, poll 6s, precision -7
Root Delay: 0.000000, Root dispersion: 2.009994, Reference-ID: 253.1.0.10
Reference Timestamp: 2587680738.000190315 (1981/12/31 19:12:18)
Originator Timestamp: 3537270946.835706873 (2012/02/03 10:15:46)
Receive Timestamp: 2587680738.000190315 (1981/12/31 19:12:18)
Transmit Timestamp: 2587680802.000191628 (1981/12/31 19:13:22)
Originator - Receive Timestamp: -949590208.835516571
Originator - Transmit Timestamp: -949590144.835515260
10:16:55.840336 arp reply 10.0.1.103 is-at e8:04:62:eb:0e:37 (oui Unknown)

Right now I am just tweaking the config file and watching the debug log on the phone hoping something will click!



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