Daniel's Intuit Blog - pulse of Asterisk & Nagios

04:31:53 pm, by novecento

, 84 words

Categories: News, Asterisk

New FOP2 - I like it... :-)

We just installed FOP 2 for a client in Los Angeles and i have to say the USD 40 spent on the license are worth every penny... :-)

Good job and greetings to Nicolás Gudiño i Argentina, the creator of this tool!

We had to tweak FreePBX a little bit to add the link into the top panel:

Edit /var/www/html/admin/views/freepbx.php

and add/change as you wish....

print_sub_tool( _("FOP-2"), "http://111.111.111.111/html/fop2/index.html" ,0, NULL, "fop2" );

Cheers
Daniel

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04:01:52 pm, by novecento

, 1538 words

Categories: News, Asterisk

Create XML based directories for Cisoc 70XX phones

Good Morning

Feroz wrote nice blog about creating XML based directories for Cisco 79XX phones. The files are stored on your Asterisk / Apache server.

Cheers
Daniel

We have successfully flash the Cisco 7941 to SIP. Now we need to create an XML files to work with Cisco 7941 SIP phones. First of all, we need to create SEP<MACADDRESS>.cnf.xml file. In our environment, we successfully connected the Cisco 7941 to Asterisk on Debian and FreePBX.

Below is a working sample of SEP<MACADDRESS>.cnf.xml file. You have to modify few parameters to reflect your environment.

<device>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>admin</sshUserId>
<sshPassword>123</sshPassword>
<devicePool>
        <dateTimeSetting>
            <dateTemplate>D/M/YA</dateTemplate>
            <timeZone>Taipei Standard Time</timeZone>
            <ntps>
                <ntp>
                    <name>your NTP server</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>your sip server name</processNodeName>
              </callManager>
           </member>
        </members>
     </callManagerGroup>
</devicePool>

<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>

<authenticationURL></authenticationURL>
<directoryURL>http://yourwebserver/dir</directoryURL>
<servicesURL></servicesURL>
<idleURL></idleURL>
<informationURL></informationURL>
<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
<dscpForCm2Dvce>96</dscpForCm2Dvce>

<certHash></certHash>
<encrConfig>false</encrConfig>
   <sipProfile>
     <sipProxies>
        <backupProxy></backupProxy>
        <backupProxyPort></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>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>
     <preferredCodec>none</preferredCodec>
     <dtmfAvtPayload>101</dtmfAvtPayload>
     <dtmfDbLevel>3</dtmfDbLevel>
     <dtmfOutofBand>avt</dtmfOutofBand>
     <alwaysUsePrimeLine>false</alwaysUsePrimeLine>
     <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
     <kpml>3</kpml>
     <stutterMsgWaiting>0</stutterMsgWaiting>
     <callStats>false</callStats>
     <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
     <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
     <startMediaPort>16384</startMediaPort>
     <stopMediaPort>32766</stopMediaPort>
     <voipControlPort>5060</voipControlPort>
     <dscpForAudio>184</dscpForAudio>
     <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
     <dialTemplate>dialplan.xml</dialTemplate>

     <phoneLabel>Display top right hand corner of the phone</phoneLabel>
     <sipLines>
        <line button="1">
           <featureID>9</featureID>
           <featureLabel>extension number</featureLabel>
           <name>extension number</name>
           <displayName>extension name</displayName>
           <contact>extension number</contact>
           <proxy>your sip server ip</proxy>
           <port>5060</port>
           <autoAnswer>
              <autoAnswerEnabled>2</autoAnswerEnabled>
           </autoAnswer>
           <callWaiting>3</callWaiting>

<authName>extension username</authName>
<authPassword>extension password</authPassword>

           <sharedLine>false</sharedLine>
           <messageWaitingLampPolicy>1</messageWaitingLampPolicy>
           <messagesNumber>*97</messagesNumber>
           <ringSettingIdle>4</ringSettingIdle>
           <ringSettingActive>5</ringSettingActive>

           <forwardCallInfoDisplay>
              <callerName>true</callerName>
              <callerNumber>false</callerNumber>
              <redirectedNumber>false</redirectedNumber>
              <dialedNumber>true</dialedNumber>
           </forwardCallInfoDisplay>
        </line>
     </sipLines>
</sipProfile>
</device>

Save the file as SEP<MACADDRESS>.cnf.xml. Note: You need to disable NAT on your phone extension. If not, your phone will not register.

Next up, we need to configure the dialplan. Below are the sample of a simple dialplan.xml that we used:

Those are the basic xml files that you needed in order to make your phone to work. You need to save the files to the same folder in your TFTP server folder as the previous post.

Once all the files are inside the folder, you would need to reboot the phone by pressing the Settings button, then **#**. The phone will reboot and upload the new SEP<MACADDRESS>.cnf.xml and the dialplan.xml files.

After it has booted up, your phone should registered and you can hear a dialtone. You can test make/receive calls.

The following post is to show you how to make Cisco IP phones display its XML based directory through its mini browser on the phones. In order for this to work, you would need a webserver (we use Apache2) to serve the phone.

Ensure your webserver interprets .xml files as text by modifying the mime types. In apache, this is done by editing the /etc/mime.types file. Add an entry

text/xml xml

In one of your apache’s web directories, say, www.yourwebserver.com/dir, create two files, menu.xml and list.xml

This below is what menu.xml should look like (modify where’s relevant)

<CiscoIPPhoneMenu>
<Title>Corporate Directory</Title>
<Prompt>Select to go get Corp Dir</Prompt>
<MenuItem>
<Name>Intuits Directory</Name>
<URL>http://yourwebserver/dir/list.xml</URL>
</MenuItem>
</CiscoIPPhoneMenu>

This below is what list.xml should look like (modify where’s relevant)

<CiscoIPPhoneDirectory>
    <Title>Intuit Directory</Title>
     <Prompt>Select To Call</Prompt>
     <DirectoryEntry>
          <Name>Feroz</Name>
          <Telephone>1006</Telephone>
     </DirectoryEntry>
     <DirectoryEntry>
          <Name>Sanjay</Name>
          <Telephone>1003</Telephone>
     </DirectoryEntry>
</CiscoIPPhoneDirectory>

Modify the <DirectoryURL> directive inside SEP<MACADDRESS>.cnf.xml (this is the phone’s configuration file) to point to http://yourwebserver/menu.xml … (example below)

<directoryURL>http://yourwebserver/dir/menu.xml</directoryURL>

Restart your phone with your TFTP server ready to serve the SEP file to your Cisco Phone.

Technorati Tags: Asterisk,Cisco,SIP,XML

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03:57:59 pm, by novecento

, 862 words

Categories: News, Asterisk

Cisco 79XX update to SIP firmware

Good Morning

Our fellow Intuiter Feroz wrote a nice blog about update Cisco 79XX phones to SIP and run them on Asterisk.

Cheers
Daniel

We have a client (MIMOS Berhad) that wanted to run Asterisk to replace their Cisco Call Manager but still use their Cisco IP Phones. We had to setup several things to ensure the Cisco IP phone to run on SIP.

This post is to show you how to flash the Cisco IP phones firmware from Cisco Skinny to SIP.

We looked everywhere but all the guides out there are not very specific and clear on how to do this.

Firstly, you need to flash your Cisco IP phones to support SIP.

This guide has been tested with Cisco IP Phone 7941 and 7911 with latest SIP firmware (8-5-2SR1s).

Before we get started, following are the items that you need:

DHCP Server
TFTP Server (i recommend to use TFTPd32 and you download it at http://tftpd32.jounin.net/)
CISCO SIP Firmware cmterm-7941_7961-sip.8-5-2SR1S.zip (you need a Cisco SMARTNET to download this from CISCO)
XMLDefault.cnf.xml

First of all, we need to extract the firmware. Create a folder on the desktop, I named it as Cisco7941. Extract the .zip file to this folder. You should see the following files:

- apps41.8-5-2CA1-6.sbn
- cnu41.8-5-2CA1-6.sbn
- cvm41sip.8-5-2CA1-6.sbn
- dsp41.8-5-2CA1-6.sbn
- jar41sip.8-5-2CA1-6.sbn
- SIP41.8-5-2SR1S.loads
- term41.default.loads
- term61.default.loads

Once we get the files sorted out, we need to create the XMLDefault.cnf.xml file to reflect our intent to reflash the phone. Below are a working sample of XMLDefault.cnf.xml:

>

t xml sorted out, we need to setup TFTP & DHCP server to serve the phone. I suggest that you connect the phone directly to your computer to avoid clashing with competing DHCP server in your network. In this case, you won’t screw up your network incase you misconfigured your TFTP and DHCP server.

Now that you have tftpd32 open and running, we need to configure it first according to our requirements.

Open the settings box. Enable DHCP Server, TFTP Server, and Syslog Server, rest can be disabled. Make sure that you have set the directory into the same directory as your Cisco Firmware Files. Set TFTP Security to STANDARD, in the image it says None. Check that the bind address is set to the correct interface as your computer LAN interface. When done, click OK to save the settings.

Then, go to the DHCP server tab. Make sure to enter the Additional Option to 150 & your TFTP server IP address. You would need to restart the tftpd32 after modifying the settings.

Now, plug in the phone and power up. Verify whether the phone receives an IP address from this DHCP server & check whether the TFTP IP address is bind to your IP (Press Settings button, Network Configuration, IPv4 Configuration and scroll until you see TFTP Server 1). If everything is correctly configured, we can start to flash the phone.

So now, it is time to reboot the phone and flash to the new SIP firmware. First, power off the phone. Before inserting the power, press the # button. Keep on pressing the # button while you insert the power, either by POE or the main power adaptor. Once the line button is flashing, release # button and the press 123456789*0# and then wait.

As the phone boots up watch the syslogs. While it is booting up, it will try to pull the SEP<MACADDRESS>.cnf then give up and move on to XMLDefault.cnf.xml. When it sees that file it should start downloading the firmware files, flashing and reboot.

After a successful reboot, it should be display a Cisco screen with the words ‘Unprovisioned’ at the bottom left. If you continue to watch syslog, you will see that it requests XMLDefault.cnf.xml, CTLSEP<MACADDRESS>.tlv and SEP<MACADDRESS>.cnf.xml on a regular basis. These files handle the provisioning of the phone, from defaults to individual per-phone settings.

You can verify whether the phone has uploaded the correct SIP firmware by going to Settings > Status > Firmware Version. The files loaded should be the same as the files listed earlier in the post.

I’ll share with you guys the working sample of the XML files needed in order to configure the phone and make it work in the next post.

Technorati Tags: Asterisk,Cisco,SIP

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05:58:32 am, by novecento

, 26 words

Categories: News

It's time for the world to unite as human race

Simply amazing song!
Im'm sure Bob is happy to see this.

Wish you all a great weekend!
One love one heart.

Cheers
Daniel

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07:14:03 am, by novecento

, 33 words

Categories: News, Asterisk

We are in the Avnet-Brochure... !!!

Avnet one of the worlds largest electronic distributor includes Intuit's Astervox solution
in it's brochure. Our Alfred Chong did a great job! Thanks.