FreePBX 12 + Cisco 7975

I have built a new FreeBPX 6.12 server and have a Cisco 7975 phone for testing

Created an extension on FreePBX
Uploaded config and to tftp
Installed OSS Endpoint Manager
Created Entry for phone MAC

What works on the phone:

  • boots
  • downloads/installs firmware
  • pulls config from FreePBX (can see getting the xml file in the phone log)
  • line and phone name appear

What does not work on the phone:

  • keeps saying registering phone
  • line has an “X” beside the phone icon

NAT is set to off as the phone and FreePBX are all on the same subnet

So i can see the phone is getting its configuration, but then whatever is meant to happen next with FreePBX to access the extension is not working

the line secret/password is correct in the config as is the line name… the line is a PJSIP line

I also tried the phone config without OSS with a manual file, using files from Cisco Call Manager as a template too and got the same result

I’ve spent around 100 hours troubleshooting this, but haven’t gotten anywhere, also the FreePBX logs are as good as useless, i can see the tftp session but nothing more from the phone trying to connect or register…

Any ideas ?

There are MANY problems with getting the 79xx phones working properly with Asterisk as they are really designed for the Cisco infrastructure and are rather Asterisk/tftp unfriendly, perhaps

http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79xx

will give you the “skinny” :wink: on how they can be persuaded to work with a SIP stack on an Asterisk based system.

Oracle,
As dicko said, the Cisco 79xx SIP phones suck.

Cisco only wants them used for Cisco PBXs.

The 7940 and 7960 are the only phones supported by Cisco TAC for third party SIP integration.

However, all other 79xx models can work with freepbx if you can downgrade the sip firmware to 8-5-4, a known good firmware.

However with the newest phones, they require ver 9 firmware or later which is a terrible firmware. Look on the bottom of your phone - what is the hardware version? Is there a sticker that says “requires ver 9 firmware or later”? If so, sell or return the phone. If not, try to downgrade to 8-5-4 (read on voip-info)

I have spent 100s of hours as well exploring trying to make 9 firmware work. I found out that the reason you cant get the phone to register is because it defaults to send SIP with TCP instead of UDP. I can help you get the phone registered by changing it to UDP, but theres no point. Once it is registered, it will have a handful of other issues that I havent figured out solutions for yet, making the phone barely even useable

Here’s how to get it registered with the 9 firmware which again, i dont recommend doing but here it is

Heres what i was talking about with it defaulting to SIP tcp

The transportLayerProtocol parameter

“where “2” is for UDP, “4” is default (which is TCP) and “1” is for TCP”

I have several Cisco 7975s connected to a FreePBX 12 (Asterisk 1.8) server. They work perfectly, but you need to apply a patch to Asterisk.
https://issues.asterisk.org/jira/browse/ASTERISK-13145

Actually, I don’t understand why the perfectly working patch isn’t implemented into Asterisk.

Here is another interesting source:
http://docs.acsdata.co.nz/asterisk-cisco/document-overview.shtml

And it is simply not true that the newest Cisco firmware does not work. I have it installed on all machines…9.4.2.1…no issues at all :wink:

Ciao
Reinhard

Reinhard,
That is great news that you have the latest SIP firmware working. Can you post an xml config for us and I will try that and the patch on my phones. Will review your sources next time I am in my lab

Thanks!

I whole heartedly disagree, the 79xx series work fine , IF correctly set up, there is so much FUD on the internet it serves to complicate things.

I have many 79xx phones all working fine with sip v9.4.x firmware, Using the latest distro there is no patching to be done, the phones work out of the gate.

Out of interest what is the secret / password you have used, the default generated one is too long and the phone truncates the password so it never matches, it therefore stays forever stuck at ‘Registering’

Please try this working XML file, this is using Chan_Sip not PJSip so u might have to change port settings.

Just copy into txt doc and rename SEPMACADDRESS.cnf.xml

Replace MACADDRESS with phones macadress :wink:

<?xml version=“1.0” encoding=“utf-8”?>
<device>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>admin</sshUserId>
<sshPassword>222222</sshPassword>
<transportLayerProtocol>2</transportLayerProtocol>
<transferonhookenabled>true</transferonhookenabled>
<stopmediaport>16399</stopmediaport>
<voipcontrolport>5061</voipcontrolport>
<rfc2543hold>true</rfc2543hold>
<calleridblocking>0</calleridblocking>
<remotepartyid>false</remotepartyid>
<devicePool>
<dateTimeSetting>
<dateTemplate>D/M/YA</dateTemplate>
<timeZone>GMT Standard/Daylight Time</timeZone>
<ntps>
<ntp>
<name>81.168.77.149</name>
<ntpMode>Unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>
<callManagerGroup>
<members>
<member>
<callManager>
<processNodeName>***SERVER IP ADDRESS </processNodeName>
<ports>
<sipPort>5061</sipPort>
</ports>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<sipProfile>
<autoAnswerTimer>1</autoAnswerTimer>
<autoAnswerAltBehavior>false</autoAnswerAltBehavior>
<autoAnswerOverride>true</autoAnswerOverride>
<transferOnhookEnabled>false</transferOnhookEnabled>
<enableVad>false</enableVad>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<phoneLabel>OT</phoneLabel>
<stutterMsgWaiting>1</stutterMsgWaiting>
<callStats>false</callStats>
<offhookToFirstDigitTimer>15000</offhookToFirstDigitTimer>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>32766</stopMediaPort>
<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate>dialplan-MACADDRESS.xml</dialTemplate>
<softKeyFile></softKeyFile>
<sipProxies>
<backupProxy>
SERVER IP ADDRESS </backupProxy>
<backupProxyPort>5061</backupProxyPort>
<emergencyProxy>
SERVER IP ADDRESS ***</emergencyProxy>
<emergencyProxyPort>5061</emergencyProxyPort>
<outboundProxy></outboundProxy>
<outboundProxyPort>5060</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>
</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>
<sipLines>
<line button=“1”>
<featureID>9</featureID>
<featureLabel>OT</featureLabel>
<proxy>USECALLMANAGER</proxy>
<port>5061</port>
<name>1029</name>
<displayName>OT</displayName>
<callWaiting>3</callWaiting>
<authName>1029</authName>
<authPassword>SECRET</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
<messagesNumber>*97</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>1029</contact>
<speedDialNumber></speedDialNumber>
<serviceURI></serviceURI>
<autoanswerenabled>2</autoanswerenabled>
<autoAnswer>
</autoAnswer>
<forwardCallInfoDisplay>
</forwardCallInfoDisplay>
</line>
</sipLines>
</sipProfile>
<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>2</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>
<webAccess>0</webAccess>
<daysDisplayNotActive></daysDisplayNotActive>
<displayOnTime>00:00</displayOnTime>
<displayOnDuration>23:59</displayOnDuration>
<displayIdleTimeout>00:10</displayIdleTimeout>
<spanToPCPort>1</spanToPCPort>
</vendorConfig>
<userLocale>
<name>English_United_Kingdom</name>
<uid></uid>
<langCode>en_US</langCode>
<version>1.0.0.0-1</version>
<winCharSet>iso-8859-1</winCharSet>
</userLocale>
<networkLocaleInfo>
<name></name>
<uid></uid>
<version>1.0.0.0-1</version>
</networkLocaleInfo>
<capfList>
<capf>
<phonePort>3804</phonePort>
</capf>
</capfList>
</device>

First thing i would check is your password, as when set up correctly these phones work very very well.

Regards

Dale

Be aware that I am using a German locale. And YES, the Asterisk patch is required, otherwise BLF won’t work.

[code]<?xml version="1.0" encoding="UTF-8"?>

true
SIP
admin
cisco


D.M.Y
W. Europe Standard/Daylight Time


131.234.137.23
unicast


192.53.103.108
unicast







192.168.0.34
asterisk

2000
5060
5061

192.168.0.34





Disable

2000

2000

2000

5060

5060

5060
false

120



true
1

SIP75.9-4-2-1S

false
false
0
1
0
0
2
1
0
0
0
0
0
1,2,3,4,5,6,7


00:05
0
0
1
1



German_Germany

de

ISO-8859-1

Austria

Austria


1
http://192.168.0.34/cisco/services/authentication.php
http://192.168.0.34/xmlservices/index2.php
0

http://192.168.0.34/xmlservices/help.php


http://192.168.0.34/xmlservices/index.php
96
0
96
4
5
0


3804



false



5060

5060

5060
true


true
x-cisco-serviceuri-cfwdall
x-cisco-serviceuri-pickup
x-cisco-serviceuri-opickup
x-cisco-serviceuri-gpickup
x-cisco-serviceuri-meetme
x-cisco-serviceuri-abbrdial
true
2
false
true
2
0
0
true


6
10
180
1200
5
120
120
5
500
4000
70
false
None

1
false
true
true
false
g711alaw
0
101
3
avt
false
false
3
0
false
10
false
16384
32766
5060
184
0
dialplan.xml
softkeys1.xml
Reinhard



9
Line 1
17
17
17
USECALLMANAGER
5060

2

1
17
xxxxxx
false
3
*9811
4
5

true
false
false
true



21
Kxxxxx
11
1


21
Exxxxx
12
1


21
Txxxx
13
1


21
Bxxxxx
14
1


21
Bxxxx
15
1


20
KAMERA
http://192.168.0.34/xmlservices/cam.php


20
HAUSTOR ÖFFNEN
http://192.168.0.34/xmlservices/open-door-17.php


[/code]

Hi,

I will add to my reply, in the context of the question asked, you do not need to patch asterisk to make the phones work, they work fine without the patch.

The patch allows the control of the BLF lamp.

Dale

Hi Dale,

without the busy lamp feature I would say that the Cisco 7975s are not fully working :wink:
It is one of the biggest advantages of the 7975 phone that you can assign up to 7 extensions and always see what those phones are doing (red, green and yellow light…and animated symbols on the screen)

Best
Reinhard

Hi Reinhard,

I totally agree, but i thought it easier to not complicate matters as there is so much mis-information about Cisco phones and asterisk.

Once the sip firmware is loaded , they are simple.

In all honesty, if you go to the trouble of using the commercial End Point manager, Cisco phones are no harder to get working than any others.

Regards

Dale

Hey everyone,

thanks for the help. I have the phone registered now, but i think ive managed to screw something else up… and incoming calls wont route to the extension, outgoing calls work, and inter office calls do not work…

here is the log from the inter office call (between a softphone and cisco 9795)

[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:1] GotoIf(“PJSIP/611-0000003f”, “1?ext-local,610,1”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Goto (ext-local,610,1)
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:1] Set(“PJSIP/611-0000003f”, “__RINGTIMER=15”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:2] Macro(“PJSIP/611-0000003f”, “exten-vm,610,610,0,0,0”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:1] Macro(“PJSIP/611-0000003f”, “user-callerid,”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:1] Set(“PJSIP/611-0000003f”, “TOUCH_MONITOR=1417412432.454”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:2] Set(“PJSIP/611-0000003f”, “AMPUSER=611”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:3] GotoIf(“PJSIP/611-0000003f”, “0?report”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:4] ExecIf(“PJSIP/611-0000003f”, “1?Set(REALCALLERIDNUM=611)”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:5] Set(“PJSIP/611-0000003f”, “AMPUSER=611”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:6] GotoIf(“PJSIP/611-0000003f”, “0?limit”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:7] Set(“PJSIP/611-0000003f”, “AMPUSERCIDNAME=611”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:8] GotoIf(“PJSIP/611-0000003f”, “0?report”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:9] Set(“PJSIP/611-0000003f”, “AMPUSERCID=611”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:10] Set(“PJSIP/611-0000003f”, “__DIAL_OPTIONS=Ttr”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:11] Set(“PJSIP/611-0000003f”, “CALLERID(all)=“611” <611>”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:12] GotoIf(“PJSIP/611-0000003f”, “0?limit”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:13] ExecIf(“PJSIP/611-0000003f”, “0?Set(GROUP(concurrency_limit)=611)”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:14] GosubIf(“PJSIP/611-0000003f”, “7?sub-ccss,s,1(macro-exten-vm,)”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:1] ExecIf(“PJSIP/611-0000003f”, “0?Return()”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:2] Set(“PJSIP/611-0000003f”, “CCSS_SETUP=TRUE”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:3] GosubIf(“PJSIP/611-0000003f”, “0?monitor_config,1(macro-exten-vm,):monitor_default,1(macro-exten-vm,)”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:1] GotoIf(“PJSIP/611-0000003f”, “0?is_exten”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:2] StackPop(“PJSIP/611-0000003f”, “”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:3] Return(“PJSIP/611-0000003f”, “FALSE”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:15] ExecIf(“PJSIP/611-0000003f”, “0?Set(CHANNEL(language)=)”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:16] GotoIf(“PJSIP/611-0000003f”, “0?continue”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:17] ExecIf(“PJSIP/611-0000003f”, “1?Set(__CALLEE_ACCOUNCODE=)”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:18] Set(“PJSIP/611-0000003f”, “__TTL=64”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:19] GotoIf(“PJSIP/611-0000003f”, “1?continue”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Goto (macro-user-callerid,s,30)
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:30] Set(“PJSIP/611-0000003f”, “CALLERID(number)=611”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:31] Set(“PJSIP/611-0000003f”, “CALLERID(name)=611”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:32] Set(“PJSIP/611-0000003f”, “CDR(cnum)=611”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:33] Set(“PJSIP/611-0000003f”, “CDR(cnam)=611”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:34] Set(“PJSIP/611-0000003f”, “CHANNEL(language)=en”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:2] Set(“PJSIP/611-0000003f”, “RingGroupMethod=none”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:3] Set(“PJSIP/611-0000003f”, “__EXTTOCALL=610”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:4] Set(“PJSIP/611-0000003f”, “__PICKUPMARK=610”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:5] Set(“PJSIP/611-0000003f”, “RT=15”) in new stack
[2014-12-01 16:40:32] WARNING[11635][C-00000045] chan_sip.c: This function can only be used on SIP channels.
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:6] ExecIf(“PJSIP/611-0000003f”, “0?Macro(vm,610,DIRECTDIAL,)”) in new stack
[2014-12-01 16:40:32] WARNING[11635][C-00000045] chan_sip.c: This function can only be used on SIP channels.
[2014-12-01 16:40:32] WARNING[11635][C-00000045] chan_sip.c: This function can only be used on SIP channels.
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:7] ExecIf(“PJSIP/611-0000003f”, “0?MacroExit()”) in new stack
[2014-12-01 16:40:32] WARNING[11635][C-00000045] chan_sip.c: This function can only be used on SIP channels.
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:8] Gosub(“PJSIP/611-0000003f”, “sub-record-check,s,1(exten,610,)”) in new stack
[2014-12-01 16:40:32] ERROR[11635][C-00000045] app_stack.c: Attempt to reach a non-existent destination for Gosub: (Context:sub-record-check, Extension:s, Priority:1)
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] app_macro.c: Spawn extension (macro-exten-vm, s, 8) exited non-zero on ‘PJSIP/611-0000003f’ in macro ‘exten-vm’
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Spawn extension (ext-local, 610, 2) exited non-zero on ‘PJSIP/611-0000003f’
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:1] Macro(“PJSIP/611-0000003f”, “hangupcall,”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:1] GotoIf(“PJSIP/611-0000003f”, “1?theend”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Goto (macro-hangupcall,s,3)
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:3] ExecIf(“PJSIP/611-0000003f”, “0?Set(CDR(recordingfile)=)”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Executing [[email protected]:4] Hangup(“PJSIP/611-0000003f”, “”) in new stack
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] app_macro.c: Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘PJSIP/611-0000003f’ in macro ‘hangupcall’
[2014-12-01 16:40:32] VERBOSE[11635][C-00000045] pbx.c: Spawn extension (ext-local, h, 1) exited non-zero on ‘PJSIP/611-0000003f’

Im still learning what all this means, but happy to share configs if you want to see what i have running so far

Hello Reinhard,

I agree with you regarding the BLF’s on the cisco phones. I have a 7970 here.
I see that you have managed to patch your FreePBX. I am curious to know how you did this because from what I understand one has to patch then install asterisk ? I believe it is not possible to patch a running system ? :pensive:
Could you explain how you achieved this patching ?
Many thanks :smiley:

Hey, I may be almost two months late, but the way I patched the official FreePBX distro with the cisco-usecallmanager patch above was to download the FreePBX SRPMS from here:

http://yum.freepbxdistro.org/pbx/SRPMS/asterisk/13/

and more or less follow this guide:

http://bradthemad.org/tech/notes/patching_rpms.php

I had to do this on a different machine than the running system. And there were a lot of dependencies that had to be installed. See here:

https://drive.google.com/file/d/0B211QsfoNBOPc3NMNHNRS3p3MGM/view

You can download my finished product and apply it to your fully updated system with one single command. More info here:

http://community.freepbx.org/t/resolved-how-do-i-apply-a-patch-to-asterisk-on-freepbx-distro/30391/11?u=grintor