Cisco Phone 7945

I have cisco phone 7945 and I have upgraded it to SIP but the phone keep registering and never seem to register with my asterisk PBX.
Kindly view my SEP(mac).cnf.xml file for any error.
<?xml version="1.0" encoding="UTF-8"?>

<device>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>admin</sshUserId>
<sshPassword>password</sshPassword>
<devicePool>
<dateTimeSetting>
<dateTemplate>D/M/Y</dateTemplate>
<timeZone>Central Standard/Daylight Time</timeZone>
<ntps>
<ntp>
<name>172.100.101.229</name>
<ntpMode>Unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5062</securedSipPort>
</ports>
<processNodeName>172.100.101.229</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<sipProfile>
<sipProxies>
<backupProxy>172.100.101.229</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>true</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>240</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>g729</preferredCodec>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<phoneLabel>LIGHTNING</phoneLabel>
<stutterMsgWaiting>1</stutterMsgWaiting>
<callStats>false</callStats>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>101</featureLabel>
<proxy>172.100.101.229</proxy>
<port>5060</port>
<name>101</name>
<displayName>101</displayName>
<autoAnswer>
<autoAnswerEnabled>10</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>101</authName>
<authPassword>line secret</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messagesNumber>*99</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>101</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
<line button="2">
<featureID>20</featureID>
<featureLabel>Menu</featureLabel>
<serviceURI>http://example.domain.ext/services/menu.xml</serviceURI>
</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>SIP45.8-5-4S</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>
<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>

Hi Godfrey,
Could you please collect packet capture from the back of the phone when it tries to register using the link:
https://supportforums.cisco.com/document/44741/collecting-packet-capture-cisco-ip-phone
If the IP phone port is not spanned, then you have to collect the sniffers from the switchport where the phone is connected to. You have to span the switch port and collect the sniffer captures
~Amit

Similar Messages

Cisco IP Phone 7945-G or 7965-G screen goes blank randomly with SCCP firmware 9.2(3) on CUCM 8.6(2a)

Hello everybody,
I have the following situation:
A WAN VoIP network, with centralized call processing model (central site and few branches)
Cisco IP Phones 7945-G and 7965-G running SCCP
Cisco Unified Communications Manager 8.6(2a)
SCCP firmware Release 9.2(3) for the above mentioned hardware models
The problem that occurs randomly is that either when the firmware was SCCP firmware Release 9.2(1), or it was upgraded to SCCP firmware Release 9.2(3), approximately 5 Cisco IP Phones, the screen goes blank, the phone being still operational (but difficult to operate in such circumstances), and the problem persists until it is disconnected from PoE, and re-connected.
Please if possible, any solution to this problem is much appreciated.
Thank you in advance.
Radu

Hello again,
In addition to what I already mentioned, I would like to detail a few aspects, in order to be more specific with this behavior:
-          From the total number of Cisco IP phones which is less than 1000, this behavior happened randomly, within 1 month and 3 weeks
-          Actually, it was a total number of 8 Cisco IP phones whose display went off, 4 being located in Headquarters, and from the other 4, 3 were in the same Branch, and 1 in another Branch
-          Also, 7 IP phones are connected in non-Cisco switches with PoE, and 1 IP Phone was power supplied from AC adapter
-          initially, all existing IP phones had firmware Release 9.2(1) for 7945G and 7965G, and the mentioned behavior happened with 4 IP phones
-          then, for all IP phones, it were upgraded to firmware Release 9.2(3) for 7945G and 7965G, but problem re-appeared with other 4 IP phones
The only solution that solved the issue for each IP phone was to disconnect the power from the IP phones.
I hope that you might help me on how to get solving this problem, and I am looking forward for any useful idea or solution.
Many thanks in advance.
Regards,
Radu

3850 - auto qos voip cisco-phone

I am having an issue where it appears that the "auto qos voip cisco-phone" command is marking down my voice packets COS and DSCP values to 0. Anyone else notice this? I am connecting Cisco 7945 phone to the interfaces with this command applied to the interface. I verified that the IP Phone is sending COS = 5 and DSCP = ef to the switchport. I verified the packets inbound to the switchport and the packets outbound as they transit to the destination IP Phone's switchport.

Hi joshua,
What is the IOS-XE version you are running on your 3850 ?
Post your switchport configuration (7945 phone connected)
Here is 3.3 QoS config guide & I noticed it says something like below qouted. So make sure this trust device command is there & your port is configured for standard VoIP port connectivity.(like voice vlan, etc)
http://www.cisco.com/c/en/us/td/docs/switches/lan/catalyst3850/software/release/3se/consolidated_guide/configuration_guide/b_consolidated_3850_3se_cg/b_consolidated_3850_3se_cg_chapter_01101000.html#reference_8A1C35AB2874498F90C4F5600301BE9D
"The trust device device_type interface configuration command is only supported in an auto-QoS configuration, and not as a stand-alone command on the switch. When using the trust device device_type interface configuration command in an auto-QoS configuration, if the connected peer device is not a corresponding device (defined as a device matching your trust policy), both CoS and DSCP values are set to "0" and any input policy will not take effect"
HTH
Rasika
***** Pls rate all useful responses ****

Cisco phone proxy will support on cucm 8.6 or not

Hi as per document i can see that Cisco phone proxy . Without using vpn connect . Customer want to keep configured Cucm on there Jabber or cisco mobile on the iphone and need to get connected when ever they have an internet access .
And also by keeping a small end router at branches havin only one cisco phone needs to connect to the corprate office using proxy.
Is that possible and also is it possible on 8.6 cucm version . Here am just attaching a document info which says version supported
Supported Cisco UCM and IP Phones for the Phone Proxy
Cisco Unified Communications Manager
The following release of the Cisco Unified Communications Manager are supported with the phone proxy:
•Cisco Unified CallManager Version 4.x
•Cisco Unified CallManager Version 5.0
•Cisco Unified CallManager Version 5.1
•Cisco Unified Communications Manager 6.1
•Cisco Unified Communications Manager 7.0

I understand the VPN solution is a good one, but we have already 100 proxy phones deployed and from what I read I'd need to first set up a VPN phone on the cluster before deploying it in the field. This would be difficult to do with users all over the country. Or am I missing something?
I have a question about our ASA though if anyone can answer it. We have to move from an ASA 5510 to a 5520 due to the 100 UC proxy-phone license limitation (even though we have 70 phones it appears that a UC license is taken up for each TFTP server configured on the phone - primary and secondary). When I configured the 5520 I can register new phones no problem. None of the existing phones will register until the CTL file is manually deleted on each phone. Is there a way to seamlessly migrate to the 5520? Such as using the same certs, CTL file etc.. on the 5520? When I configured it it generated it's own self-signed key and CTL file. I did import the certificates from CUCM...

Cisco Phone Control and Presence 8.6.1.1185 with IBM Lotus Notes 8.5.2 (Integrated Sametime Client 8.0.2) - No presence status visible

Hi community,
I am trying to integrate Cisco Unified Presence 8.6.1.10000-34 with IBM Lotus Notes 8.5.2 with the integrated Sametime Client version 8.0.2 via the Cisco Plugins 8.6.1.1185.
Phone control is working fine, whereas the presence status is not shown (= no handset symbol next to the Sametime user). When I look in the preferences of the plugin, I can see that the plugin has connected successfully to the CUCM (8.6.2.20000-2),whereas the connection to the CUPS has not been established.
The user id as well as the password are all the same on all systems. Here is a description of what I have configured via the ciscocfg.exe tool:
Feature Control:
- Enable Phone Status -> checked
- Enable Dial Using Cisco IP Communicator -> unchecked (not required)
- Enable Control Desk Phone -> checked
- Default Mode -> Control Desk Phone
Control Desk Phone Settings:
- Voicemail Pilot Number -> left blank (no voicemail)
- Cisco Unified Communications Manager
     - Servers -> IP address of CUCM
     - Read Only -> unchecked
     - Use as Default CUCM -> checked
     - Synchronize Credentials -> checked
          - Use Sametime Credentials -> checked
Use Secure Connection: -> not required
LDAP Phone Attributes: -> not required
Phone Status Settings:
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- Read Only -> unchecked
- Synchronize Credentials -> checked
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- Sametime User ID Mapping
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Thanks a lot in advance!
Kind regards,
Igor

Hi all,
here are some additions to my above post:
Servers and clients used:
1x CUCM 8.6.2.20000-2
1x CUPS 8.6.1.10000-34
1x IBM Lotus Domino Messaging Express Server 8.5.2
1x Sametime Entry Server 8.5.2 (on top of the Domino server)
2x IBM Lotus Notes 8.5.2 with integrated Sametime 8.0.2
2x Cisco Phone Control and Presence with Lotus Sametime (PCAP) 8.6.1.1185
2x Cisco Unified Personal Communicator 8.5.5.19839
Setup:
- CUCM, CUPS and CUPC are working fine, i.e. Desk Phone control via CUPC, as well as availability and presence status are working without issues
- IBM Lotus Domino server is the LDAP Directory, the Sametime Entry Server is installed on top of the Domino server and uses the Domino Directory
- User ID and password on CUCM/CUPS match the ShortName field and password in Domino
- The PCAP plug-in has been manually deployed to both Notes clients with the following configuration:
     - Enable Phone Status -> active
     - Desk Phone Control -> active
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     - Sametime User ID Mapping is implemented via the LDAP Attribute uid (which is equal to the user id in CUCM)
     - LDAP configuration filled in with details of the Domino server
Phone Control is working fine, also the connection to the LDAP server (Domino) is fine. However, when I type in the credentials for the CUPS server login, I can see (in Troubleshooting pane) that the user (pparker) is connected to the CUPS server for a short period of time and then gets disconnected. After that no connection is possible to the CUPS server, i.e. status is always disconnected.
I have collected the Tomcat (EPASSoap00010.log and security00010.log) logs via RTMT and compared them to the logs from the PCAP plugin. The relevant time period is from 15:14 to 15:17. In the Tomcat logs I can see that the authentication is successful (see attached files), however in the log of PCAP plugin I can see the following messages:
2012/02/03 15:14:35.281 WARNUNG Credential is rejected. Nothing to retry ::class.method=com.cisco.sametime.phonestatus.cup.CUPPresenceWatcher.answerChallenge() ::thread=CT_CALLBACK.1 ::loggername=com.cisco.sametime.phonestatus.cup
2012/02/03 15:14:35.281 WARNUNG #### Connection rejected presence server ::class.method=com.cisco.sametime.phonestatus.cup.CUPPresenceWatcher.onPresenceServerConnectionRejected() ::thread=CT_CALLBACK.1 ::loggername=com.cisco.sametime.phonestatus.cup
2012/02/03 15:14:35.281 WARNUNG Credential is rejected. Nothing to retry ::class.method=com.cisco.sametime.phonestatus.cup.CUPPresenceWatcher.answerChallenge() ::thread=CT_CALLBACK.2 ::loggername=com.cisco.sametime.phonestatus.cup
2012/02/03 15:14:35.281 WARNUNG #### Connection rejected presence server ::class.method=com.cisco.sametime.phonestatus.cup.CUPPresenceWatcher.onPresenceServerConnectionRejected() ::thread=CT_CALLBACK.2 ::loggername=com.cisco.sametime.phonestatus.cup
I don't understand why the connection is rejected although the Sametime Internal ID and CUPS User ID match. Does anyone know what the issue could be?
All posts are very much appreciated!
Thanks a lot in advance!
Kind regards,
Igor

Cisco 2960x - auto qos voip cisco-phone on access ports

After configuring "auto qos voip cisco-phone" on a stack of 2960x switches, we later noticed that after a power cycle (config was saved first) it went missing when doing a "show run" from the interface configs, on one of the switches (g1/0/1-48 had it applied, then after power cycle it didn't show). However, doing a "show auto qos interface" shows it is still applied to all interfaces g10/1-48.
I try to do "auto qos voip cisco-phone" again on g1/0/1-48, and it shows this:
SwitchStack01(config-if-range)#auto qos voip cisco-phone
AutoQoS Error: AutoQoS already configured
% Range command terminated because it failed on GigabitEthernet1/0/1
If I attempt to remove "auto qos voip cisco-phone", it shows this:
SwitchStack01(config-if-range)#no auto qos voip cisco-phone
AutoQoS Error: AutoQoS not configured
% Range command terminated because it failed on GigabitEthernet1/0/1
When I do a "show run", it is no longer displayed for the first switch of the stack.So is it still applied to the interfaces or not? Is this a bug? Has anyone else experienced this? This has happened to at least a couple stacks of 2960x's we have.
Thanks!

It appears to be a bug in the code. Open a TAC ticket with Cisco and send them the output and the IOS version you are using so they can help you resolve the issue. Most likely, they recommend an upgrade.
HTH

"click to call" posibillity in cisco phones ?

Hello
Is there any way to call numbers on cisco phones directy from clik on website ?
I am looking for sollution like in skype :
https://support.skype.com/en/faq/FA12264/what-is-the-skype-add-on
Users install add on on computer webbrowsers and numbers are automicaly highlited and then user can click this and call
I am looking this for hardware voip phones.
Is this possible on any cisco or non-cisco phones ?
Sorry for my bad english

Skype is working to provide a Click to Call update as close to a Firefox browser update release as possible. The current Skype click to call plugin is compatible with Firefox 26.
Follow the latest Skype Community News
↓ Did my reply answer your question? Accept it as a solution to help others, Thanks. ↓

Call manager 5 in cisco phone

Hello All,
Maximum 3 call managers can be added in CM group and one SRST reference. So totally 4 CM servers but why we have CM server 5 in all cisco phones under network configuration.
-Murali

perhaps the plan at one point was to support more, and nobody ever bothered to adjust the page. In either case 3 +1 as you state is the max.
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Auto qos voip cisco-phone

Hello,
I am very new to voip, second day. I have configured a 3850 stack to trust cisco phones on the switchports with the auto qos feature. I am interested to know where else I have to apply qos for the policy's to apply. The 3850 stack is uplinked to a 4500x "core". My question is once the qos is applied at the switch port, is everything just grand and my qos will work or does it have to be applied elsewhere.
Thank you.
Dallas

It has to be applied all throughout your network. Ie Core switch, access layer, uplinks to voice servers/gateways and access ports to phones. If you have wireless phones, those will have to be considered as well.

Cisco Phone Directory - Max 32 DirectoryEntry object

Cisco SPA525G
3CX Ver 11 Windows 7 Pro IIS
We are able to access the Corporate Phone directory from the Cisco phones. However, it lists only the first 32 phone contacts.
Cisco documentations states:
http://www.cisco.com/en/US/docs/voice_i ... #wp1033491
"A single CiscoIPPhoneDirectory object can contain a maximum of 32 DirectoryEntry objects. If more than 32 entries must be returned, use multipleCiscoIPPhoneDirectory objects in subsequent HTTP requests."
3CX only maintains one XML file for all contacts found in the 3CX system. Ours has over 50 contacts listed in the file.
This explains why the Cisco phones successfully downloads only the first 32 objects found in the file and the rest are discarded.
Possible Solution:
Create a program which when run, reads and breaks the 3CX Corporate Directory XML file into a series of alternative XML files each containing only 32 objects each.
This I understand…
However, I am confused on how to implement Cisco’s method of “…subsequent HTTP requests."
Based in the information below, I believe that somehow I am to add to the HTTP response a “HTTP Refresh Setting” with the time parameter = 0, and the URL path of the next XML file that should follow in the directory listing.
Can some explain or point me to documentation on how to do this?
Thanks,
David
http://docstore.mik.ua/univercd/cc/td/d ... .htm#25866
HTTP Refresh Setting
The HTTP headers sent with any page from an HTTP server can include a Refresh setting. This setting has two parameters: a time in seconds and a URL. These two parameters direct the recipient to wait the time given in the seconds parameter, then get the data pointed to by the URL.
The Cisco IP Phone HTTP client properly supports this setting, which gives a great deal of power to service developers. It means that any XML object being displayed can be replaced with a new page after a fixed amount of time.
Figure 8 shows a good example of how the refresh setting can be used. This sample page shows the user the current value of Cisco's stock. It first brings up a splash screen that displays the Yahoo logo. After a very short time, it displays the numeric Cisco stock parameters. Finally, it shows a graph of Cisco's intra-day stock performance. The display then repeatedly cycles between the final two views.
Figure 8 Sample Refresh Display
All of this can occur without user intervention, as the display automatically cycles if a timer parameter is specified. However, on any given screen, the user can force an immediate reload by pressing the Update soft key. Also, if a timer parameter of 0 was sent in the header, the page will never automatically reload. In that case, the display will only move to the next page when the Update soft key is pressed. If no refresh URL is specified, the current page will be reloaded.
MIME Type and Other HTTP Headers
Delivering pages with the proper MIME type and other formatting items is not difficult, but it requires moderately in-depth knowledge of your web server. The following code excerpt, written in JavaScript and used with Microsoft's IIS and ASP, sets these values in a few lines:
<%@ Language=JavaScript %>
<%
Response.AddHeader( "Refresh",
"3; url=http://services.cisco.com/s/q.asp");
Response.ContentType = "text/xml";
. Additional page content here
%>
Usually, you can set the MIME type for pages in any web server by simply performing an association to the .xml file extension. Your web server's documentation should explain how to accomplish this. This allows you to serve static pages without the need for writing script.
If you want to deliver dynamic content using the other supported HTTP headers, you will need to understand how to generate the HTTP headers using the desired programming language and have CGI or script access on the target web server.
Content Expiration Header Setting
The expiration header can control what URLs are added to the phone's URL history. This is slightly different behavior than traditional web browsers but is implemented to perform the same function. Disable the back button functionality to avoid calling a URL twice.
This functionality allows you to expire the content of any page that is sent to the phone. When a user presses the Exit soft key, they are taken back to the last URL that was not expired when it was loaded. This differs from traditional browsers by not considering the current freshness of the data but the freshness of the data when the URL was requested. This requires you to expire a page when it is first loaded and to not set a time and date in the future.
The following example shows how to expire content on IIS using Active Server Page (ASP):
<%@ Language=JavaScript %>
<%
Response.ContentType = "text/xml";
Response.Expires = -1;
%>
The "Expires" property is the number of minutes to wait to expire the content. Setting this value to -1 subtracts 1 minute from the request time and returns a date and time that has already passed.
DBPlus2
New User
Posts: 8
Joined: Thu Apr 05, 2012 5:45 pm

It seems you are asking in wrong forum. AFAIK, you are asking "how to add HTTP header to response generated by my own script". It depend on WWW server we are speaking of and language of script itself. If you will fail to found solution within documentation of the HTTP server and/or scripting language you are using, then the better place for your question is a forum related to such language and HTTP server.
In meantime, you can try other solituin. The "Refresh: 0;..." header is required for correct function of SoftKey:Next which is displayed by default. But you can redefine the content of SoftKey area using your own key. Such configuration is part of DirectoryObject you sent to phone. See definition of SoftKey 3 in example bottom. It's not original SoftKey:Next that depend on Refresh header. It's my own custom soft-key named "Next" with exact URL defined as part of key definition (replace 'N' with number of next page). It doesn't depend on Refresh header in any way. You should consider such advice as "temporary workaround". You should discover how to send HTTP header 'Refresh' from your script. Note, it's not possible to redefine one SoftKey only. If you wish to redefine a soft-key, then all soft-keys need's to be defined by you.
... followed by Title, Prompt,up to 32 ...
Dial
SoftKey:Dial
1
EditDial
SoftKey:EditDial
2
Next
https://an-url-to-your-server-and-script/test-Directory.asp?page=N
3
Cancel
SoftKey:Cancel
4
Exit
SoftKey:Exit
5

Last out of services information on Cisco phone

Does anyone got any informaton of the meaning on the code of ReasonForOutOfService from the debug display on the Cisco Phone web page?

I have the same problem, but I'm see this ReasonForOutOfService 18 on CP-9951 SIP Phones.
Has anyone more information of this case or a workaround for it ?
Also, I can see more other Reason in the same IP Phone how bellow:
09:59:49 ReasonForOutOfService=18
10:15:52 ReasonForOutOfService=12
10:17:53 ReasonForOutOfService=18
10:45:27 ReasonForOutOfService=12
10:45:29 ReasonForOutOfService=100
21:00:39 ReasonForOutOfService=0
11:14:52 ReasonForOutOfService=23
It is a historical of ReasonForOutServices on last 24 hours on the same IP Phone.

Catalyst Express 500 802.1q with non-Cisco Phones

This weekend we spent hours trying to get 802.1q tagging to work on a VLAN with ShoreTel phones. The user interface on this switch seems to only allow "Cisco-Voice" VLAN, without any specifics. This didn't work. The specs on this switch say that the .1q is supported, but we couldn't figure it out. The more expensive switches were easier to configure for Voip QoS.
Can anyone advise me on the tricks to getting this to work with the lower end Catalyst Express 500? Or does this switch only support 802.1q with Cisco phones?

Cisco IP Phone uses CDP to let the ip phone know what vlan it's suppose to be (via voice-vlan). shore tel would definitely not use CDP since CDP is cisco proprietory, so it's voice vlan must be defined on it, I rememer Avaya being the same way. So, having said that, just make sure that the Shore tel Ip phone are in the right vlan. what does not work anyway? shore Tel IP Phone will not come up? Will not get it's configuration from it's software PBX? Use the smartport configuration on CE500.
Please rate all posts.

Cisco Phone 7960 and SIP provider

Hi,
i have an account with a Sip provider.
I have all information for make a connection with xlite sip client but if i try to configure a Cisco Phone with SIP Firmware (7.5), phone not work.
My provider is messagenet.it.
Can you help me?
Thanks

Hello,
have a look at the configuration guide "Getting Started with Your Cisco SIP IP Phone" at
http://www.cisco.com/en/US/products/sw/voicesw/ps2156/products_administration_guide_chapter09186a0080080edf.html
This should pretty much answer your questions and allow you to succeed with your task.
Hope this helps! Please rate all posts.
Regards, Martin

Cdr doesn't bill Cisco phone 6901 Call Manager release 9

Hello Engineers,
Actually I opened a TAC for a problem that I got in my customer's environment.
I have CUCM release 9 and Cisco phone 6901 - firmware: SCCP6901.9-3-2.loads
When I tried to see the calls from CAR I didn't get output from the CAR/CDR.
I did an upload to the firmware SCCP6901.9-3-1-SR1-3.loads.
After that all calls were ok.
I hope this post can help anothers engineers

Hi Karina,
Thanks for taking time out of your busy schedule to help others
here @ CSC Much appreciated!
Cheers!
Rob
"Seek it out and ye shall find "
- OneRepublic

Cisco phone 7906

Dears,
I have problem with cisco phone 7906 , that phone it is working good suddenly it give ( ip configuring and registring )
and the phone not working ( hang in ip configuring and registring ) when i try to make upgrade it is hange ,
when i try to but ip address manually it is also hange , and the phone not working .
please can you help me .
Thanks,

Hi Mohammad,
Sorry to say the IP Phone 7906G is not supported in any 4.1(2) version :(
Cisco Unified IP Phone 7906G Requires Cisco Unified CallManager release 5.0(3) or later; 4.2(1)SR1 or later; 4.1(3)SR3a or later; or 3.3(5)SR2 or later.
From this good doc;
http://www.cisco.com/en/US/products/hw/phones/ps379/prod_release_note09186a008068f542.html
Hope this helps! I know its a bummer.
Rob

Cisco Phone 7945

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