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saviopereira1
Beginner

Configuring Cisco CP-7821 sip

Hi,

  I am trying to set up Cisco CP-7821 with a third party Hosted VoIP service provider but I can't just find a way to configure the phone, we already have SPA504g phones set up with the same Hosted VoIP service provider. There is no CUCM or any other Cisco Call manager involved.

Please find the phone information below;

Phone Model: Cisco CP-7821

Active Load: sip78xx.10-2-1-12SR1-4

Is it possible to set this phone manually to work with third party sip services ? 

Thanks in advance,

Sav

15 REPLIES 15
Leo L
VIP Community Legend

Post the "dialplan.xml" file. 

Post the SEP<MAC address of the phone>.cnf.xml file.

Hi Leo,

     Please find the files as an attachment. I've just changed the exts from .xml to .txt

Sav

The second one doesn't look like valid .cnf file. It seems to be MS Word document. I suspect the 7821 will not understand it.

Hi Dan/Leo,

Can you please have a look at the new .xml file & i have also attached a copy of TFTP log file, I reckon that something is wrong. 

Kind Regards,

Sav

I'm no expert on this class of devices, thus I can't tell the content of sep....cnf is correct - but it's format looks good now.

Also I can provide just generic comment related to tfto log. The phone is asking for CTLSEPBCC49396F15C.tlv and  ITLSEPBCC49396F15C.tlv. Both  doesn't exist.

SEPBCC49396F15C.cnf.xml is downloaded successfully but /sp-sip.jar as well as United_States\g3-tones.xml are nonexistent.

I don't know neither what's the purpose of files missing nor they're required.

I wish that Leo or other expert on the matter will provide you more specific response.

 

Leo L
VIP Community Legend

JAR, TLV & tones files are not important when dealing with non-Cisco call-manager (like Asterisk).   

Tell me why I do not have a menu in addition?
This phone has a Web Access?
The manual says that it's all there, but I can not find it.
Cisco cp-7821-k9 firmware SIP 11.0.1
Leo L
VIP Community Legend

I don't understand the question.

In the description on the phone there is a menu item "advanced"

Admin Login > advanced >

But in the menu on the phone, I can not find it.

Leo L
VIP Community Legend

<?xml version="1.0" encoding="UTF-8"?>
<device>
    <versionStamp>{7821 Aug 28 2015 12:40:48}</versionStamp>
    <devicePool>
        <dateTimeSetting>
            <dateTemplate>D/M/YA</dateTemplate>
            <timeZone>GMT</timeZone>
        </dateTimeSetting>
        <callManagerGroup>
            <members>
                <member  priority="0">
                    <callManager>
                        <ports>
                            <ethernetPhonePort>2000</ethernetPhonePort>
                        </ports>
                        <processNodeName>192.168.1.100</processNodeName>
                    </callManager>
                </member>
            </members>
        </callManagerGroup>
    </devicePool>
    <commonProfile>
        <callLogBlfEnabled>3</callLogBlfEnabled>
    </commonProfile>
    <loadInformation>sip78xx.10-2-1-12SR1-4</loadInformation>
    <userLocale>
        <name></name>
        <langCode>en</langCode>
    </userLocale>
    <networkLocale>United_States</networkLocale>
    <networkLocaleInfo>
        <name>United_States</name>
    </networkLocaleInfo>
    <idleTimeout>0</idleTimeout>
    <authenticationURL></authenticationURL>
    <directoryURL></directoryURL>
    <idleURL></idleURL>
    <informationURL></informationURL>
    <messagesURL></messagesURL>
    <proxyServerURL></proxyServerURL>
    <servicesURL></servicesURL>
    <capfAuthMode>0</capfAuthMode>
    <capfList>
        <capf>
            <phonePort>5060</phonePort>
            <processNodeName></processNodeName>
        </capf>
    </capfList>
    <deviceSecurityMode>1</deviceSecurityMode>
</device>

Errrr ... The "format" of the SEP<macaddress>.cnf.xml is correct but the contents are missing a lot of lines of information.   The phone won't be able to join the VoIP call manager because the above lines does NOT specify the username/password to authenticate to.  

See below: 

<sipLines>
  <line button="1" lineIndex="1">
    <featureID>9</featureID>
    <featureLabel>FEATURE LABEL</featureLabel>
    <name>NAME</name>
    <displayName>DISPLAY NAME</displayName>
    <contact>CONTACT</contact>
    <proxy>USECALLMANAGER</proxy>
    <port>5060</port>
    <autoAnswer>
    <autoAnswerEnabled>2</autoAnswerEnabled>
    </autoAnswer>
    <callWaiting>3</callWaiting>
    <authName>USERNAME</authName>
    <authPassword>PASSWORD</authPassword>
    <sharedLine>false</sharedLine>
    <messageWaitingLampPolicy>3</messageWaitingLampPolicy>
    <messageWaitingAMWI>1</messageWaitingAMWI>
    <messagesNumber>VOICE-MAIL NUMBER</messagesNumber>
    <ringSettingIdle>4</ringSettingIdle>
    <ringSettingActive>5</ringSettingActive>
    <forwardCallInfoDisplay>
    <callerName>true</callerName>
    <callerNumber>false</callerNumber>
    <redirectedNumber>false</redirectedNumber>
    <dialedNumber>true</dialedNumber>
    </forwardCallInfoDisplay>
    <maxNumCalls>4</maxNumCalls>
    <busyTrigger>2</busyTrigger>
  </line>
<sipLines>

Do NOT, under any circumstances remove or change the line "proxy>USECALLMANAGER</proxy>".   All other XML tags in CAPs (in BLUE) you need to fill up.

 

Leo L
VIP Community Legend

<processNodeName>192.168.1.100</processNodeName>

Is this the IP address of the hosted VoIP call server?  The <callManagerGroup> is missing a few lines as well.  

        <callManagerGroup>
            <tftpDefault>true</tftpDefault>
            <members>
                <member priority="0">
                    <callManager>
                        <ports>
                            <ethernetPhonePort>2000</ethernetPhonePort>
                            <sipPort>5060</sipPort>
                            <securedSipPort>5062</securedSipPort>
                        </ports>
                        <processNodeName>ASTERISK SERVER IP ADDRESS</processNodeName>
                    </callManager>
                </member>
            </members>
        </callManagerGroup>

Hello, please tell me after the line is added the <sipLines>

Hi, how would the configuration for 2 lines in the model 7821 be? Thank you

I use this as a example, but I stuck on registration in progress, I update the script using my PBX configuration,

I have Elastix 2.0 the SIP Trunk it on Puerto Rico and PBX it in Miami, I am using right now SP504g and we are upgrade to 7821.

 

can I use this IP Phone without SSH login, I saw a lot of person using scripts that include the SSH, I don't need this, I saw this one but I don't have any idea why it still stuck on registration progress, other its about CLT and ITL

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