Hi Dan/Leo,Can you please

saviopereira1 · ‎08-25-2015

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

Leo Laohoo · ‎08-26-2015

Post the "dialplan.xml" file.

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

saviopereira1 · ‎08-26-2015

Hi Leo,

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

Sav

Dan Lukes · ‎08-26-2015

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.

saviopereira1 · ‎09-01-2015

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

Dan Lukes · ‎09-01-2015

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 Laohoo · ‎09-01-2015

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

antydrchert · ‎09-16-2016

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 Laohoo · ‎09-16-2016

I don't understand the question.

antydrchert · ‎09-16-2016

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 Laohoo · ‎09-01-2015

<?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 Laohoo · ‎09-01-2015

<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>

emalvarez@oesia.com · ‎09-28-2017

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

rtizolr · ‎05-24-2018

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

jdelacruz90 · ‎02-13-2019

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

Configuring Cisco CP-7821 sip