Sharpen Astesrisk to work with Cisco BLF
BLF (Busy Lamp Field) - in general, it is a set of indicators on an IP phone that show the status of a line: free or busy.
In this article, we will teach Asterisk to work properly with Cisco phones to display BLF statuses.
Let's enter the command in the Asterisk console:
The State column shows, in fact, the state of the line, Watchers - the number of people listening to this line. When the state of the line changes, Asterisk sends a message to the telephone and turns on the telephone. The performance of BLF on Asterisk can first be checked on a soft phone, for example, on DrayTek SoftPhone. After adding the contacts, you can see the indication of the busy line. By default, Asterisk sends such messages that the Cisco phone cannot read, and, accordingly, the BLF does not work for them.
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Downloading the patch: issues.asterisk.org/jira/secure/attachment/32739/cisco-blf-asterisk.1.8.0.patch
Installing this patch will not compile asterisk.
There is a small typo in the patch, it needs to be fixed.
In line 23, switch (state) is corrected to switch (data-> state) ;
in line 40 of the state, correct to data-> state ;
in line 60 state correct for data ;
in line 66 state corrected to data .
Next, copy the Asterisk source to a folder, for example / usr / src / asterisk.
We put the patch in the channels folder and immediately rename it, for example in patch.
Execute the command:
On the request “File to patch”, enter the path where sip.h is located (usually in include / sip / sip.h).
If Asterisk is installed, you must remove it. To do this, stop Asterisk:
Remove Asterisk:
Before installing asterisk, you need to install all the modules for compiling sources, in particular gcc, gcc-c ++, kernel-devel, ncurses-devel, openssl-devel, m2crypto, libssl-devel.
Next, perform:
Run:
On this with Asterisk done.
Consider the basic lines for BLF
featureID 21 means that you must use BLF for this Directory Number.
In the /etc/asterisk/sip_general_additional.conf file, add or modify, if any, the lines:
The following devices were used to organize BLF:
Phones: Cisco 7961, 7962, FreePBX 2.0.2, Asterisk 1.8.11.
In this article, we will teach Asterisk to work properly with Cisco phones to display BLF statuses.
Theory
Let's enter the command in the Asterisk console:
asterisk*CLI> core show hints -= Registered Asterisk Dial Plan Hints =- 1196@ext-local : SIP/1196 State:Ringing Watchers 3 1188@ext-local : SIP/1188 State:InUse Watchers 3 5030@ext-local : SIP/5030 State:Idle Watchers 1 The State column shows, in fact, the state of the line, Watchers - the number of people listening to this line. When the state of the line changes, Asterisk sends a message to the telephone and turns on the telephone. The performance of BLF on Asterisk can first be checked on a soft phone, for example, on DrayTek SoftPhone. After adding the contacts, you can see the indication of the busy line. By default, Asterisk sends such messages that the Cisco phone cannot read, and, accordingly, the BLF does not work for them.
')
Patching Asterisk source codes
Downloading the patch: issues.asterisk.org/jira/secure/attachment/32739/cisco-blf-asterisk.1.8.0.patch
Installing this patch will not compile asterisk.
There is a small typo in the patch, it needs to be fixed.
In line 23, switch (state) is corrected to switch (data-> state) ;
in line 40 of the state, correct to data-> state ;
in line 60 state correct for data ;
in line 66 state corrected to data .
Next, copy the Asterisk source to a folder, for example / usr / src / asterisk.
We put the patch in the channels folder and immediately rename it, for example in patch.
Execute the command:
patch chan_sip.c patch On the request “File to patch”, enter the path where sip.h is located (usually in include / sip / sip.h).
If Asterisk is installed, you must remove it. To do this, stop Asterisk:
service asterisk stop Remove Asterisk:
yum remove asterisk Before installing asterisk, you need to install all the modules for compiling sources, in particular gcc, gcc-c ++, kernel-devel, ncurses-devel, openssl-devel, m2crypto, libssl-devel.
Next, perform:
./configure make menuselect -- make make install Run:
asterisk start On this with Asterisk done.
Configure phone files config
config.file
<?xml version="1.0" encoding="UTF-8"?> <device xsi:type="axl:XIPPhone" ctiid="94"> <fullConfig>true</fullConfig> <deviceProtocol>SIP</deviceProtocol> <sshUserId>cisco</sshUserId> <sshPassword>cisco</sshPassword> <devicePool> <dateTimeSetting> <dateTemplate>YMD</dateTemplate> <timeZone>Central Asia Standard Time</timeZone> <ntps> <ntp> <name>PBX-IP Address</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>PBX-IP Address</processNodeName> </callManager> </member> </members> </callManagerGroup> <srstInfo> <srstOption>Disable</srstOption> <ipAddr1></ipAddr1> <port1>2000</port1> <ipAddr2></ipAddr2> <port2>2000</port2> <ipAddr3></ipAddr3> <port3>2000</port3> <sipIpAddr1></sipIpAddr1> <sipPort1>5060</sipPort1> <sipIpAddr2></sipIpAddr2> <sipPort2>5060</sipPort2> <sipIpAddr3></sipIpAddr3> <sipPort3>5060</sipPort3> <isSecure>false</isSecure> </srstInfo> <connectionMonitorDuration>120</connectionMonitorDuration> </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-cisco-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> <retainForwardInformation>false</retainForwardInformation> </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>true</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> <kpml>3</kpml> <alwaysUsePrimeLine>false</alwaysUsePrimeLine> <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail> <phoneLabel></phoneLabel> <stutterMsgWaiting>2</stutterMsgWaiting> <callStats>true</callStats> <offhookToFirstDigitTimer>15000</offhookToFirstDigitTimer> <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts> <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig> <poundEndOfDial>false</poundEndOfDial> <startMediaPort>16384</startMediaPort> <stopMediaPort>32766</stopMediaPort> <sipLines> <line button="1" lineIndex="1"> <featureID>9</featureID> <proxy>USECALLMANAGER</proxy> <port>5060</port> <autoAnswer> <autoAnswerEnabled>2</autoAnswerEnabled> </autoAnswer> <callWaiting>3</callWaiting> <sharedLine>false</sharedLine> <messageWaitingLampPolicy>3</messageWaitingLampPolicy> <messagesNumber></messagesNumber> <ringSettingIdle>4</ringSettingIdle> <ringSettingActive>5</ringSettingActive> <forwardCallInfoDisplay> <callerName>true</callerName> <callerNumber>false</callerNumber> <redirectedNumber>false</redirectedNumber> <dialedNumber>true</dialedNumber> </forwardCallInfoDisplay> <!-- ***************************** --> <featureLabel>LineName 5037</featureLabel> <displayName>LineName</displayName> <name>5037</name> <authName>5037</authName> <authPassword>password</authPassword> </line> <line button="2"> <featureID>21</featureID> <featureLabel>Description</featureLabel> <speedDialNumber>Directory Number</speedDialNumber> </line> </sipLines> <externalNumberMask></externalNumberMask> <!-- ***************************** --> <voipControlPort>5060</voipControlPort> <dscpForAudio>184</dscpForAudio> <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy> <dialTemplate>dialplan.xml</dialTemplate> </sipProfile> <commonProfile> <phonePassword></phonePassword> <backgroundImageAccess>true</backgroundImageAccess> <callLogBlfEnabled>1</callLogBlfEnabled> </commonProfile> <vendorConfig> <disableSpeaker>false</disableSpeaker> <disableSpeakerAndHeadset>false</disableSpeakerAndHeadset> <forwardingDelay>1</forwardingDelay> <pcPort>0</pcPort> <settingsAccess>1</settingsAccess> <garp>0</garp> <voiceVlanAccess>0</voiceVlanAccess> <videoCapability>0</videoCapability> <autoSelectLineEnable>0</autoSelectLineEnable> <webAccess>1</webAccess> <daysDisplayNotActive>1,7</daysDisplayNotActive> <displayOnTime>09:00</displayOnTime> <displayOnDuration>12:00</displayOnDuration> <displayIdleTimeout>01:00</displayIdleTimeout> <spanToPCPort>1</spanToPCPort> <loggingDisplay>2</loggingDisplay> <loadServer>10.1.96.31</loadServer> <recordingTone>0</recordingTone> <recordingToneLocalVolume>100</recordingToneLocalVolume> <recordingToneRemoteVolume>50</recordingToneRemoteVolume> <recordingToneDuration></recordingToneDuration> <displayOnWhenIncomingCall>0</displayOnWhenIncomingCall> <rtcp>0</rtcp> <moreKeyReversionTimer>5</moreKeyReversionTimer> <autoCallSelect>1</autoCallSelect> <logServer>10.1.96.31</logServer> <g722CodecSupport>0</g722CodecSupport> <headsetWidebandUIControl>0</headsetWidebandUIControl> <handsetWidebandUIControl>0</handsetWidebandUIControl> <headsetWidebandEnable>0</headsetWidebandEnable> <handsetWidebandEnable>0</handsetWidebandEnable> <peerFirmwareSharing>0</peerFirmwareSharing> <enableCdpSwPort>1</enableCdpSwPort> <enableCdpPcPort>1</enableCdpPcPort> </vendorConfig> <versionStamp>1200501729-ee9247c4-1a10-481c-8fdc-612737c5aadd</versionStamp> <deviceSecurityMode>1</deviceSecurityMode> <idleTimeout>0</idleTimeout> <authenticationURL></authenticationURL> <directoryURL></directoryURL> <idleURL></idleURL> <informationURL></informationURL> <messagesURL></messagesURL> <proxyServerURL></proxyServerURL> <servicesURL></servicesURL> <dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig> <dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices> <dscpForCm2Dvce>96</dscpForCm2Dvce> <transportLayerProtocol>2</transportLayerProtocol> <singleButtonBarge>0</singleButtonBarge> <capfAuthMode>0</capfAuthMode> <capfList> <capf> <phonePort>3804</phonePort> <processNodeName>PBX-ip-address</processNodeName> </capf> </capfList> <certHash></certHash> <encrConfig>false</encrConfig> <advertiseG722Codec>1</advertiseG722Codec> </device> Consider the basic lines for BLF
<line button="2"> <featureID><b>21</b></featureID> <featureLabel>Description</featureLabel> <speedDialNumber>Directory Number</speedDialNumber> </line> featureID 21 means that you must use BLF for this Directory Number.
In the /etc/asterisk/sip_general_additional.conf file, add or modify, if any, the lines:
allowsubscribe=yes notifyhold=yes callcounter=yes notifyringing=yes limitonpeers=yes buggymwi=yes Used devices
The following devices were used to organize BLF:
Phones: Cisco 7961, 7962, FreePBX 2.0.2, Asterisk 1.8.11.
Source: https://habr.com/ru/post/159783/
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