Discussion: Asterisk phone cisco 79xx

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Sip configuration manager for Cisco 79xx configuration Files

I'm making a configuration manager for Cisco phones to make easier the management of configuration files. If anyone is interested please check my git: https://github.com/xpheres/SipManagerSuite

by xpheres, Monday 03 of August, 2015 (13:39:06 UTC)
Cisco 79XX SIP Firmware

Contact me at scott@srellis.com for the Cisco SIP Firmware if you don't have it, I also have the samples of .cnf config files you need, they are working .cnf files... ones that I used that included various functions for about 10 phones

by mrsrellis, Friday 16 of August, 2013 (07:24:00 UTC)
Perl script for Cisco 7960 VoIP phone to dial number remotely over Telnet.

the script that will dial a number and hung up after number of seconds specified.
EXAMPLE1: perl scriptname.pl 172.16.31.10 303 5 lim
this would dial number 303 on cisco phone @172.16.31.10 and hung up after 5 seconds.
EXAMPLE2: perl scriptname.pl 172.16.31.10 303
this would dial number 303 on cisco phone @172.16.31.10 and exit.


# check for no arguments; if none act like 'cat'
if ($#ARGV == -1) {
	print <"1st IP\n2nd number to dial\n3ed seconds to talk\n\n4th type lim if u want to limit conversation\n">; exit;

} elsif ($ARGV[0] eq "-h") {
	print <"help\n">;
}


$ip=$ARGV[0];
$number=$ARGV[1];
$slp=$ARGV[2];
$stay=$ARGV[3];

use Net::Telnet;
sub login {
$telnet = new Net::Telnet ( Timeout=>10, Errmode=>'die');
$telnet->open("$ip");
$telnet->waitfor('/Password :$/i');
$telnet->print('cisco');
$telnet->waitfor('/admin> $/i');
}

sub dial_num {
if (status() eq "IDLE") {
print $telnet->cmd('test open');
print $telnet->cmd('test key spkr');
print $telnet->cmd("test key $number");
print $telnet->cmd('test key soft3');
}
	else {
print "BUSY, phones state=$state\n";
}	
	}


sub status {
@lines = $telnet->cmd('show lsm');
$lsm = @lines[4];
$state =  substr $lsm, 19, 4;
return ($state);
}

sub hungup {
if (status() eq "IDLE") {
exit;
}
else {print $telnet->cmd('test key soft2');}
}



login();
if (status() ne "IDLE") {
exit;
}
else {
dial_num();
if ($ARGV[3] eq lim) {
sleep($slp);
hungup();}
else {
exit
}
}



by norim, Monday 07 of April, 2008 (13:34:08 UTC)
Cisco IP Phone Visual Voicemail

I have written a Visual Voicemail script for the Cisco IP Phone connected to Asterisk. It has been tested with Asterisk 1.4 and a Cisco 7970 IP Phone. It is written in Perl. You can find it on my minimal website: http://norrisnet.homeip.net
Feel free to contact me at cory.norris@earthlink.net with comments, concerns and suggestions.


What this script does is gives you a way to access, play, and delete your voicemails all from the Cisco IP Phone's screen/interface.

The specifics are this:
When you add the URL listed in the example services.xml file to your services menu for your Cisco Phone, it invokes the script. The script queries the phone to obtain the phone's extension. It uses that extension as the mailbox ID for the Asterisk/trixbox voicemail system. It then checks the Asterisk voicemail directory structure under your voicemail id and displays a list of available sub folders (Inbox and other unique ones). This list of folders is displayed on the phone's screen. You then may select a sub folder. When done, the script displays a list of voicemails within that directory providing the full caller id and timestamp. If you highlight a message (in any order) and select "Play" from the phone's menu, the message will play over the phone's speaker in the same fashion that, say, a ringtone would. If you highlight a message and press the "Delete" button, the message will be deleted.

Some specifics are that the menus are "CiscoIPPhoneGraphicFileMenu" types which are not supported on all Cisco IP phones. I plan to release a newer version that uses a menu type that is compatible with a given Cisco IP Phone model when oyu provide the model number.

Additionally, for the file to play, I use SoX to create a .raw sound file from Asterisk's .wav. I create it in the /tftpboot directory with the timestamp as the name. I am still working on a reliable mechanism to clean-up these .raw files after the message is played.

I also have this written for configurations where the web server providing the script access, the tftp server providing the phone's config and the Asterisk server are all running on the same machine.

by fcnorris4, Tuesday 13 of November, 2007 (20:26:09 UTC)
7941g and ad hoc conferences

We are using 7941g's (SIP41.8-2-2SR1S) and having some trouble with ad hoc conferences. The conferencing works fine but we can't drop off individual legs. Looking at Cisco we should have a "remove" softkey when the conference is going on but don't, only the "EndCall" key. Highlighting any number in the conference then hitting it ends the entire conference as expected Any ideas why the "remove" key doesn't show up? Or where the softkeys are stored/enabled/disabled?

by spyder40, Monday 30 of April, 2007 (17:35:37 UTC)
7941g and ad hoc conferences

We are using 7941g's (SIP41.8-2-2SR1S) and having some trouble with ad hoc conferences. The conferencing works fine but we can't drop off individual legs. Looking at Cisco we should have a "remove" softkey when the conference is going on but don't, only the "EndCall" key. Highlighting any number in the conference then hitting it ends the entire conference as expected Any ideas why the "remove" key doesn't show up? Or where the softkeys are stored/enabled/disabled?

by spyder40, Monday 30 of April, 2007 (17:30:34 UTC)
New features on 797X an SCCP 8.2.2

Hello,

on the 8.2.2 there are some new features (e.g. <displayOnWhenIncomingCall>), but how to configure it in xml?
Does somebody have an example for the new xml-file.

The <displayOnWhenIncomingCall> can be putted under <vendorConfig> with 0/1 as values

by Chaos2000, Wednesday 11 of April, 2007 (13:48:25 UTC)
Cisco 7961

Hi Guyz,

I am almost dead working on 7961. Anyone here got working one? in tcpdump I cannot see the errors. It's communicating but never gets registered. Anyone here knows what's happening.
I have used
cmterm-7941_7961-sip.8-2-2SR1 for firmware upgrade.

What is wrong? Any idea?
Thanks for reading.

my conf file is here SEP-MAC-.conf.xml


<device>

<deviceProtocol>SIP</deviceProtocol>

<sshUserId>root</sshUserId>

<sshPassword>cisco</sshPassword>

<devicePool>
<dateTimeSetting>
<dateTemplate>D-M-YA</dateTemplate>
<timeZone>+270</timeZone>
<ntps>
<ntp>
<name>172.16.33.30</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>172.16.33.30</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<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>
g711ulaw
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<natEnabled>0</natEnabled>
<natAddress></natAddress>
<phoneLabel>Charmed</phoneLabel>
<stutterMsgWaiting>2</stutterMsgWaiting>
<callStats>false</callStats>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBu
rsts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>32766</stopMediaPort>
<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>465</featureLabel>
<proxy>172.16.33.30</proxy>
<port>5060</port>
<name>465</name>
<displayName>465</displayName>
<autoAnswer>
<autoAnswerEnabled>1</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>465</authName>
<authPassword>123</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
<messagesNumber>465</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>465</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
<line button="2">
<featureID>21</featureID>
<featureLabel>465</featureLabel>
<speedDialNumber>465</speedDialNumber>
</line>
</sipLines>
<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate>DRdialplan.xml</dialTemplate>
</sipProfile>
<commonProfile>
<phonePassword>cisco</phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>
<loadInformation>SIP41.8-0-2SR1S</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>1</webAccess>
<spanToPCPort>1</spanToPCPort>
<loggingDisplay>1</loggingDisplay>
<loadServer></loadServer>
</vendorConfig>
<versionStamp>1143565489-a3cbf294-7526-4c29-8791-c4fce4ce4c37</versionStamp>
<networkLocale>us</networkLocale>
<networkLocaleInfo>
<name>us</name>
<version>5.0(2)</version>
</networkLocaleInfo>
<deviceSecurityMode>0</deviceSecurityMode>
<authenticationURL>http://www/authenticate.php</authenticationURL>
<directoryURL>http://www/directory.xml</directoryURL>
<idleURL></idleURL>
<informationURL>http://www/GetTelecasterHelpText.jsp</informationURL>
<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<servicesURL>http://www/services.xml</servicesURL>
<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>


by AstroGuru, Thursday 05 of April, 2007 (13:44:15 UTC)
Re: MWI with SIP 8.3

What does he means with "the extension/SIP ID configured in asterisk" ? The sip username ?

by yansolo90, Wednesday 17 of January, 2007 (15:33:25 UTC)
7941 with SIP 8.4 firmware

Here is a working xml SEPXXX.cnf.xml example. Everything works except MWI (bad notify error). Also, the phone reported "Erro Updating Locale" if you look at the log but everything seems fine. Please note: to enable web access, you have to set "0" on <webAccess>0</webAccess> in the xml. The following example assumed that you already have the phone upgraded with SIP firmware. The "loadInformation" was lefted blank by purpose.

<device>

<deviceProtocol>SIP</deviceProtocol>

<sshUserId>cisco</sshUserId>
<sshPassword>cisco</sshPassword>

<devicePool>
<dateTimeSetting>
<dateTemplate>M/D/Ya</dateTemplate>
<timeZone>Eastern Standard/Daylight Time</timeZone>
<ntps>
<ntp>
<name>yourpbxip</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>yourpbxip</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>

<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>

<loadInformation></loadInformation>

<vendorConfig>
<disableSpeaker>false</disableSpeaker>
<disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
<pcPort>0</pcPort>
<settingsAccess>1</settingsAccess>
<garp>1</garp>
<voiceVlanAccess>0</voiceVlanAccess>
<videoCapability>0</videoCapability>
<autoSelectLineEnable>0</autoSelectLineEnable>

<webAccess>0</webAccess>
<spanToPCPort>1</spanToPCPort>
<loggingDisplay>1</loggingDisplay>
<loadServer></loadServer>
</vendorConfig>
<userLocale>
<name>English_United_States</name>
<uid>1</uid>
<langCode>en</langCode>
<version>4.1(3)</version>
<winCharSet>iso-8859-1</winCharSet>
</userLocale>
<networkLocale>United_States</networkLocale>

<networkLocaleInfo>
<name>United_States</name>
<uid>64</uid>
<version>4.1(3)</version>
</networkLocaleInfo>

<deviceSecurityMode>1</deviceSecurityMode>

<authenticationURL>http://yourpbxip/cisco/services/authentication.php</authenticationURL>
<directoryURL>http://yourpbxip/cisco/services/PhoneDirectory.php</directoryURL>
<idleURL></idleURL>
<informationURL>http://yourpbxip/cisco/services/help.php</informationURL>

<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<servicesURL>http://yourpbxip/cisco/services/index_cisco.php</servicesURL>
<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>

<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>
none
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>

<natEnabled>0</natEnabled>
<natAddress></natAddress>

<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>phone-label-next-to-clock-no-dash-or-space</phoneLabel>
<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>Line 1</featureLabel>
<name>extension-number</name>
<displayName>displayname-noshowup-on-phone</displayName>
<contact>7b452e87-4496-4762-e11f-b26751a1884b</contact>

<proxy>yourpbxip</proxy>
<port>5060</port>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>

<authName>extension-number</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>


by betatester, Thursday 30 of November, 2006 (19:12:25 UTC)