G722 codec on Cisco 7800 series

FreePBX Endpoints
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1

I have a Cisco 7821 phone, and I configured it to connect to freepbx, it works but only on G711u codec. I would like to use G722 codec. Could someon help me? any help appreciated.
This is my SEPmac.cnf.xml

<device>
    <fullConfig>true</fullConfig>
    <deviceProtocol>SIP</deviceProtocol>
    <devicePool>
        <dateTimeSetting>
            <dateTemplate>D/M/YY</dateTemplate>
            <timeZone>Central Europe Standard/Daylight Time</timeZone>
            <ntps>
                <ntp>
                    <name>0.pl.pool.ntp.org</name>
                    <ntpMode>Unicast</ntpMode>
                </ntp>
            </ntps>
        </dateTimeSetting>
        <callManagerGroup>
            <tftpDefault>true</tftpDefault>
                <members>
                <member priority="0">
                <callManager>
                <name>192.168.1.16</name>
                <description>Station</description>
                <ports>
                  <ethernetPhonePort>2000</ethernetPhonePort>
                  <sipPort>5060</sipPort>
                  <securedSipPort>5061</securedSipPort>
                </ports>
                <processNodeName>192.168.1.16</processNodeName>
                </callManager>
                </member>
                </members>
             </callManagerGroup>
    </devicePool>
    <commonProfile>
        <phonePassword></phonePassword>
        <backgroundImageAccess>true</backgroundImageAccess>
        <callLogBlfEnabled>1</callLogBlfEnabled>
    </commonProfile>
    <vendorConfig>
        <disableSpeaker>false</disableSpeaker>
        <disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
        <pcPort>0</pcPort>
        <settingsAccess>1</settingsAccess>
        <garp>0</garp>
        <voiceVlanAccess>0</voiceVlanAccess>
        <videoCapability>1</videoCapability>
        <autoSelectLineEnable>0</autoSelectLineEnable>
        <daysDisplayNotActive>1,2,3,4,5,6,7</daysDisplayNotActive>
        <displayOnTime>00:01</displayOnTime>
        <displayOnDuration>00:01</displayOnDuration>
        <displayIdleTimeout>00:01</displayIdleTimeout>
        <webAccess>1</webAccess>
        <spanToPCPort>1</spanToPCPort>
        <loggingDisplay>1</loggingDisplay>
        <loadServer></loadServer>
    </vendorConfig>
    <deviceSecurityMode>1</deviceSecurityMode>
    <idleTimeout>0</idleTimeout>
    <directoryURL></directoryURL>
    <servicesURL>$SERVICESURL</servicesURL>
    <idleURL></idleURL>
    <messagesURL></messagesURL>
    <proxyServerURL></proxyServerURL>
    <dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
    <dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
    <dscpForCm2Dvce>96</dscpForCm2Dvce>
    <transportLayerProtocol>2</transportLayerProtocol>
    <capfAuthMode>0</capfAuthMode>
    <capfList>
        <capf>
            <phonePort>3804</phonePort>
        </capf>
    </capfList>
    <certHash></certHash>
    <encrConfig>false</encrConfig>
    <sipProfile>
        <sipProxies>
            <backupProxy></backupProxy>
            <backupProxyPort></backupProxyPort>
            <emergencyProxy></emergencyProxy>
            <emergencyProxyPort></emergencyProxyPort>
            <outboundProxy></outboundProxy>
            <outboundProxyPort></outboundProxyPort>
            <registerWithProxy></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>1</dndControl>
	<dndCallAlert>5</dndCallAlert>
        <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>
        <preferredCodec>g722</preferredCodec>
       <dtmfAvtPayload>101</dtmfAvtPayload>
       <dtmfDbLevel>3</dtmfDbLevel>
       <dtmfOutofBand>avt</dtmfOutofBand>
        <alwaysUsePrimeLine>false</alwaysUsePrimeLine>
        <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
        <kpml>3</kpml>
        <stutterMsgWaiting>1</stutterMsgWaiting>
        <callStats>true</callStats>
        <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
        <disableLocalSpeedDialConfig>true</disableLocalSpeedDialConfig>
        <startMediaPort>10100</startMediaPort>
        <stopMediaPort>10300</stopMediaPort>
        <voipControlPort>5060</voipControlPort>
        <dscpForAudio>184</dscpForAudio>
        <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
        <dialTemplate>dialplan.xml</dialTemplate>
        <phoneLabel>Ext. 2100</phoneLabel>
          <natReceivedProcessing>false</natReceivedProcessing>
          <natEnabled>false</natEnabled>
          <natAddress></natAddress>
        <sipLines>
          <line button="1">
            <featureID>9</featureID>
            <featureLabel>Ext. 2100</featureLabel>
            <proxy>192.168.1.16</proxy>
            <port>5060</port>
            <name>2100</name>
            <displayName>2100</displayName>
            <autoAnswer>
              <autoAnswerEnabled>2</autoAnswerEnabled>
            </autoAnswer>
            <callWaiting>3</callWaiting>
            <authName>2100</authName>
            <authPassword>12345678</authPassword>
            <sharedLine>false</sharedLine>
            <messageWaitingLampPolicy>1</messageWaitingLampPolicy>
            <messagesNumber>*97</messagesNumber>
            <ringSettingIdle>4</ringSettingIdle>
            <ringSettingActive>5</ringSettingActive>
            <contact>2100</contact>
            <forwardCallInfoDisplay>
              <callerName>true</callerName>
              <callerNumber>false</callerNumber>
              <redirectedNumber>false</redirectedNumber>
              <dialedNumber>true</dialedNumber>
            </forwardCallInfoDisplay>
          </line>
          <line button="2">
             <featureID>130</featureID>
             <featureLabel>Do Not Disturb</featureLabel>
          </line>
        </sipLines>
    </sipProfile>
    <phoneServices>
     <provisioning>2</provisioning>
        <phoneService  type="1" category="0">
        <name>Missed Calls</name>
        <url>Application:Cisco/MissedCalls</url>
        <vendor></vendor>
        <version></version>
        </phoneService>
        <phoneService  type="2" category="0">
                <name>Voicemail</name>
                <url>Application:Cisco/Voicemail</url>
                <vendor></vendor>
                <version></version>
        </phoneService>
        <phoneService  type="1" category="0">
                <name>Received Calls</name>
                <url>Application:Cisco/ReceivedCalls</url>
                <vendor></vendor>
                <version></version>
        </phoneService>
        <phoneService  type="1" category="0">
                <name>Placed Calls</name>
                <url>Application:Cisco/PlacedCalls</url>
                <vendor></vendor>
                <version></version>
        </phoneService>
        <phoneService  type="1" category="0">
                <name>PhoneBook</name>
                <url>http://192.168.1.16/cisco/menu.xml</url>
                <vendor></vendor>
                <version></version>
        </phoneService>
    </phoneServices>
</device>
2

@tmittelstaedt is the forum expert on Cisco stuff. Perhaps he will respond.

3

I got it working

<device>
    <fullConfig>true</fullConfig>
    <deviceProtocol>SIP</deviceProtocol>
    <devicePool>
        <dateTimeSetting>
            <dateTemplate>D/M/YY</dateTemplate>
            <timeZone>Central Europe Standard/Daylight Time</timeZone>
            <ntps>
                <ntp>
                    <name>0.pl.pool.ntp.org</name>
                    <ntpMode>Unicast</ntpMode>
                </ntp>
            </ntps>
        </dateTimeSetting>
        <callManagerGroup>
            <tftpDefault>true</tftpDefault>
                <members>
                <member priority="0">
                <callManager>
                <name>192.168.1.16</name>
                <description>Station</description>
                <ports>
                  <ethernetPhonePort>2000</ethernetPhonePort>
                  <sipPort>5060</sipPort>
                  <securedSipPort>5061</securedSipPort>
                </ports>
                <processNodeName>192.168.1.16</processNodeName>
                </callManager>
                </member>
                </members>
             </callManagerGroup>
    </devicePool>
    <commonProfile>
        <phonePassword></phonePassword>
        <backgroundImageAccess>true</backgroundImageAccess>
        <callLogBlfEnabled>1</callLogBlfEnabled>
    </commonProfile>
    <vendorConfig>
        <disableSpeaker>false</disableSpeaker>
        <disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
        <pcPort>0</pcPort>
        <settingsAccess>1</settingsAccess>
        <garp>0</garp>
        <voiceVlanAccess>0</voiceVlanAccess>
        <videoCapability>1</videoCapability>
        <autoSelectLineEnable>0</autoSelectLineEnable>
        <daysDisplayNotActive>1,2,3,4,5,6,7</daysDisplayNotActive>
        <displayOnTime>00:01</displayOnTime>
        <displayOnDuration>00:01</displayOnDuration>
        <displayIdleTimeout>00:01</displayIdleTimeout>
        <webAccess>1</webAccess>
        <spanToPCPort>1</spanToPCPort>
        <loggingDisplay>1</loggingDisplay>
        <loadServer></loadServer>
        <preferredCodec>g722</preferredCodec>
        <g722CodecSupport>2</g722CodecSupport>
    </vendorConfig>
    <advertiseG722Codec>1</advertiseG722Codec>
    <deviceSecurityMode>1</deviceSecurityMode>
    <idleTimeout>0</idleTimeout>
    <directoryURL></directoryURL>
    <servicesURL>$SERVICESURL</servicesURL>
    <idleURL></idleURL>
    <messagesURL></messagesURL>
    <proxyServerURL></proxyServerURL>
    <dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
    <dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
    <dscpForCm2Dvce>96</dscpForCm2Dvce>
    <transportLayerProtocol>2</transportLayerProtocol>
    <capfAuthMode>0</capfAuthMode>
    <capfList>
        <capf>
            <phonePort>3804</phonePort>
        </capf>
    </capfList>
    <certHash></certHash>
    <encrConfig>false</encrConfig>
    <sipProfile>
        <sipProxies>
            <backupProxy></backupProxy>
            <backupProxyPort></backupProxyPort>
            <emergencyProxy></emergencyProxy>
            <emergencyProxyPort></emergencyProxyPort>
            <outboundProxy></outboundProxy>
            <outboundProxyPort></outboundProxyPort>
            <registerWithProxy></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>1</dndControl>
	<dndCallAlert>5</dndCallAlert>
        <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>
        <preferredCodec>g722</preferredCodec>
        <dtmfAvtPayload>101</dtmfAvtPayload>
        <dtmfDbLevel>3</dtmfDbLevel>
        <dtmfOutofBand>avt</dtmfOutofBand>
        <alwaysUsePrimeLine>false</alwaysUsePrimeLine>
        <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
        <kpml>3</kpml>
        <stutterMsgWaiting>1</stutterMsgWaiting>
        <callStats>true</callStats>
        <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
        <disableLocalSpeedDialConfig>true</disableLocalSpeedDialConfig>
        <startMediaPort>10100</startMediaPort>
        <stopMediaPort>10300</stopMediaPort>
        <voipControlPort>5060</voipControlPort>
        <dscpForAudio>184</dscpForAudio>
        <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
        <dialTemplate>dialplan.xml</dialTemplate>
        <phoneLabel>Ext. 2100</phoneLabel>
          <natReceivedProcessing>false</natReceivedProcessing>
          <natEnabled>false</natEnabled>
          <natAddress></natAddress>
        <sipLines>
          <line button="1">
            <featureID>9</featureID>
            <featureLabel>Ext. 2100</featureLabel>
            <proxy>192.168.1.16</proxy>
            <port>5060</port>
            <name>2100</name>
            <displayName>2100</displayName>
            <autoAnswer>
              <autoAnswerEnabled>2</autoAnswerEnabled>
            </autoAnswer>
            <callWaiting>3</callWaiting>
            <authName>2100</authName>
            <authPassword>12345678</authPassword>
            <sharedLine>false</sharedLine>
            <messageWaitingLampPolicy>1</messageWaitingLampPolicy>
            <messagesNumber>*97</messagesNumber>
            <ringSettingIdle>4</ringSettingIdle>
            <ringSettingActive>5</ringSettingActive>
            <contact>2100</contact>
            <forwardCallInfoDisplay>
              <callerName>true</callerName>
              <callerNumber>false</callerNumber>
              <redirectedNumber>false</redirectedNumber>
              <dialedNumber>true</dialedNumber>
            </forwardCallInfoDisplay>
          </line>
          <line button="2">
             <featureID>130</featureID>
             <featureLabel>Do Not Disturb</featureLabel>
          </line>
        </sipLines>
    </sipProfile>
    <phoneServices>
     <provisioning>2</provisioning>
        <phoneService  type="1" category="0">
        <name>Missed Calls</name>
        <url>Application:Cisco/MissedCalls</url>
        <vendor></vendor>
        <version></version>
        </phoneService>
        <phoneService  type="2" category="0">
                <name>Voicemail</name>
                <url>Application:Cisco/Voicemail</url>
                <vendor></vendor>
                <version></version>
        </phoneService>
        <phoneService  type="1" category="0">
                <name>Received Calls</name>
                <url>Application:Cisco/ReceivedCalls</url>
                <vendor></vendor>
                <version></version>
        </phoneService>
        <phoneService  type="1" category="0">
                <name>Placed Calls</name>
                <url>Application:Cisco/PlacedCalls</url>
                <vendor></vendor>
                <version></version>
        </phoneService>
        <phoneService  type="1" category="0">
                <name>PhoneBook</name>
                <url>http://192.168.1.16/cisco/menu.xml</url>
                <vendor></vendor>
                <version></version>
        </phoneService>
    </phoneServices>
</device>
4

Are you sure that it works? The Ciscos usually need a Cisco-patch applied to Asterisk, because they are not fully compatible…e.g. BLF will not work.

https://usecallmanager.nz/documentation-overview.html

5

I’m going to admit, im too stupid to patch asterisk, I just don’t know how to do it from that website (I have encountered it like 2 days ago). I think it works, when I go to admin settings > status > call statistics. It says g722

6

besides, the latest patch is for asterisk 20, i already installed freepbx 17 which comes with asterisk 21. i can reinstall, but how do I make sure it installs asterisk 20?

7

It’s not worth doing it. Old Cisco phones are great hardware but the SIP-firmware is usually full of bugs.
I still use around 10 Cisco 8961s with a freePBX16 system and a patched Asterisk…they work 100%…but this is an old installation. I would not do it again.
If money isn’t an issue, buy a couple of Sangoma P370 phones.

8

well, I like cisco phones, and 7821 I bought were literally 7$ a piece in bulk so I think it’s worth tinkering some to get them working. AND I already have them here, sooo

If anyone is able to help me patch asterisk, I would apreciate that

9

I forgot…no the P370s arent good either, because they are relatively complicated to provision with freePBX. You need a certificate and https etc…
There are other SIP-compatible phones around, but Cisco is really a bad choice.
e.g. do you have a phonebook on your Cisco? No? It is really hard to get it work. :wink:

10

I already bought the cisco phones, 15 of them. they are working without asterisk patched but with some features missing, that would probably go away once asterisk is patched. I have not come far in configuring freepbx so I can reinstall and start over. I don’t really want to buy different phones since I already have them here and they work well as they are, just would be nice if they worked better.

11

Well, there is an Asterisk-switch in freePBX. You have to downgrade to version 20. (I think the option exists)
If you are not an experienced user, you will break freePBX…

12

Which version contains asterisk 20? or 18? either will work, on debian or not. It will work

13

This is the threat you are looking for…I dont know, if it works

14

I spent the whole night trying to do this and got an error every time. I give up man. It works fine as is I guess.

15

It took me weeks…so one night is nothing. :wink:
Does Cisco still provide the SIP-firmware files for these phones? I dont think so. There is a reason, why the phones are so cheap. :wink:

1 Like
16

Hi Buba,

Cisco 78xx phones DO NOT need the Cisco USECALLMANAGER patch unless you want BLF keys and/or multiple line appearance buttons on the same phone. My employer has around 300 extensions on a Cisco brand UCM and EIGHT of the extensions in the entire company have BLF keys and multiple extensions - and 4 of those are in the IT department!!! BLF is a requirement for certain setups but NOT for others, and a great many organizations don’t have ANY phones that are setup like this.

Note that for the Cisco 78xx phones that are hardware version 3 or later you can buy a firmware update and license from Cisco for about $45 a phone that will convert them to “multiprotocol AKA 3PCC” firmware in which case they then act exactly like a Linksys SPA or Polycom or any other VoIP phone out there from the standpoint of multiple extensions and BLF keys.

Absolutely nothing is wrong with running a site of a bunch of these phones in so-called “enterprise firmware” mode on chan_pjsip channel driver and just converting a few extensions over to 3PCC firmware if you must have BLF on those.

The USECALLMANAGER patch will apply to Asterisk 21 and later in conjunction with the chan_sip driver from interlinkd1. I’ve done it and have just been waiting for the author of USECALLMANAGER to get around to releasing it since I don’t want to steal his thunder - but I’m getting very disgusted with the situation, the constant ghosting from them when I email them instructions on how to combine, etc. Apparently, he and his patch have some grudge against the maintainer of chan_sip and it appears the feeling is mutual. Both claim that the USECALLMANAGER patch cannot apply to the current chan_sip but I have an Asterisk 21 PBX setup that is running it.

But getting back to the phones themselves, there is way too much focus on the idea that you -need- the Cisco patch on Asterisk. You do not. Nothing is wrong with buying cheap inexpensive used Cisco phones and using a dumptruck load of them to populate your desktops and then for the odd border case (like a receptionist station, etc.) where you need multiple extensions, getting a different brand of phone.

Unfortunately, the developers of chan_pjsip designed it so that supporting BLF on the Enterprise firmware on the Cisco phones is non-trivial. Doesn’t mean it can’t be done, just means that doing it would require a lot of development time, and the world is full of used Polycom Soundpoints, VVXes, and so on that can be had for $20 or under per phone - so the work likely will never be done and we will continue to see growing numbers of VoIP desk phones on the used market.

What is happening with high volume call centers is they are all HEAVILY pushing use of softphones on PCs. We are looking at outsourcing just our call center and the last interview I had with a potential vendor I almost had to get into a yelling match over the phone with them before they would quit pushing the softphone and grudgingly admit that yes, we could register a deskphone into their system.

I kind of get the feeling looking at the Fortune 500 that they are all holding on to phone systems they bought years ago that are all using hardware desk phones and fighting the move to the cloud. Every once in a while one of them capitulates which then causes hundreds or thousands of used deskphones to be dumped on to the secondary market. But most are being pushed into softphones on desktops by their cloud vendors and the cloud PBX vendors are doing this to lower the cost of entry so that dumping an on-prem PBX does not incur a giant capital expense in phones.

The situation is very fluid right now in used deskphones. If you got 15 78xx phones at $7 ea that’s a good deal - I’m sitting on dozens of 7821’s that we are replacing with 8845’s and it would probably cost me $7 a phone just in shipping them so you basically got them for free IMHO. I’m also sitting on close to 100 Cisco 6921’s that will run SIP code and register to Asterisk but I cannot give them away at this point. (they are on Ebay right now and have been for months)

Getting a phone book and the directory button to work on these phones is trivial, BTW. But the Cisco phones are limited to 35 entries. Beyond that you need to setup an LDAP server. But that is also the same problem with the Polycom’s as well, and with many other deskphones. In the corporation it is generally a non-issue because people can look at a vcard in their email system to find an internal extension so the phone directory isn’t used much on the phone.

1 Like
17

The deal is that I bough these phone for my dad, he’s a manager in a small local factory and they rely on walkie talkie PMR radios (Here in poland PMR is essentially the same as FSR in the US) basically license free radios, but with all the heavy machinery that has 3 phase motors and what not, they encounter a lot of interference. Now with that being said I like to tinker with this kind of stuff so I told him i could do something like a PBX system on a budget, it doesn’t even tie in to the outside world, and what i’ve done with the config i posted above it works, so I don’t see why i need to apply the patch. I am, though, thankful and grateful for the community’s response and helping me. I wish nothing but the best for all you and Merry Christmas everyone!

18

I see…yes, for this purpose it is ok. There are other low cost solutions around, which might be a better choice.
What internet router do you use? There is e.g. Fritzbox (AVM). Some models have a built in phone system, a phonebook, a voicemail. You can connect all sorts of phones to it, sip-phones, dect-phones, isdn-phones, analogue phones…even smartphones (wifi).
And it works out-of-the-box…very reliable!

19

ubiquiti edgerouter X sfp, internet coming on the sfp port fiber adapter from orange. 2 subnets one for cameras and one for computers, everything custom set up by me.

20

Do not spent waisted time in creating any sip-mac.xml…whatever file to feed the phone with it. I did, but finally gave up. As long as it works, keep it as it is. If you like to have a web-config-server on the phone, and want to use some of the embedded features (like BLF as said above) you need to change the firmware of the phone to 3PCC (the opposite of what you currently have). I bought 3 licences for my CP-8841 (30 Euros each) which are now working on 3PCC. However still no phonebook (to read from contact manager of the PBX), where e.g. yealink can php-read nearly live read it. Anyway buying such 3PCC licences in Germany (and maybe in Poland, too) is a very challenging task. Simply to find an authorized distributor consumed month. Cisco’s website to activate the licence, once you got one, is also everything else than self explaining. But worked.

Finally, if you are a cisco-fan - such as myself for design reasons - I would advice you to some old SPA514 or SPA525 phone (refurbished) which have all needed features, can be configured by embedded web-browser, understand BLF and have support for PBX-phonebook via hosted side on /var/www/html/xxxx and look look like good old Streets-of SanFancisco

1 Like