UPDATE:
Thanks to your answares i managed to google a lot and find out what to do!
I’m sooo thankfull man, really appreciated!
Regards!
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Remove stand-by on closed lid laptop
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Audio is not good
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AMI permision denied Asterisk
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Search on UCP
Thx!
Issue here is that the freepbx is used by an outgoing call center and so we are talking about thousand of calls per phone line. Sorting will not help as there will be too many pages to go through.
I will open a feature request.
have a nice day!
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Re-route calls based on sip peer availability
thanks for the replies guys, interesting solutions although I’m not sure how reliable the delayed follow me would be in this situation. Honestly, I’m not even sure how to incorporate a delayed follow me into a ring group.
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Audio is not good
If conferencing is the only purpose of this PBX, or if your meetings will have more than about six people, I suggest using a commercial conference service instead. There are some decent free ones and excellent inexpensive ones; I’ll provide recommendations if you are interested.
For troubleshooting, first report the simplest case that has bad audio. If possible, use an IP phone with a wired Ethernet connection for these tests. Next best is a softphone on a PC with a wired connection and a good headset.
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From an internal extension, call *43 (echo test). Listen carefully to the long announcement preceding the test (the actual test is not useful here).
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Enable voicemail on an extension. In Voicemail Advanced Settings, turn on Review message for the extension. Call the extension, let it go to voicemail and record a message. Press # then 2 to review the message, listen carefully for problems.
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Call directly from one extension to another and check for voice quality problems.
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Temporarily route your incoming number to an extension and have someone call in. If you have problems with quality, report whether it affects what the extension user hears, what the remote party hears or both.
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Enable music on hold for your conference. Call in from only one extension and listen for problems with the music.
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Have a second internal user call into the conference and check for audio issues.
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Have more users call in until the sound goes bad. Report the number of users when you first have trouble. Also tell us the number of users you would like to support.
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Pick up calls from queues
Hello everyone
I know that there are already some post with kinda same question, but no one got a real answare, i cant find out anything usefull by jut googling and i didnt find anything on the wiki so here I am.
essentially, I just want to know if it is possible to pick up calls from a queue!
here is an example:
Lets says that A call my receptionist (wich is B) asking to talk to C
B call C wich is busy! now, since B must be sure that C want to accept the caller, B need to put A in a wait/hold/pause mode!
I’m aware of the “parking” functionality but it doesnt fit so well what i want to achieve
I need that B can put A in a queue to itself with a minimumn time of waiting around 5 minutes!
this in order to let B try to call back C and check if it still busy or its free, or just take others incoming calls!
but i need that, if C is free, B can pick up the call from the queue so A wont wait the full 5 min!
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Using Reverse Proxy breaks Ajax - ajaxRequest declined - Referrer - GET /admin/ajax.php?command=authping HTTP/1.1" 403 43
Hello,
I fresh installed 14.0.5.2 on a CentOS7 system with a dedicated WAN ip with the help of page (newbies are not allowed to add website links)
Connecting to the WAN ip over http (port 80) works flawless. Just like others , i found some sites, i would like to use a reveser proxy. So a client connects to an FQDN freepbx.WEBSITE (notice the https) this is the reverse proxy with an other WAN ip as the freepbx CentOS host. The Reverse proxy forwards the request to the CentOS WAN ip over http 80. CentOS only accepts request from the reverse proxy IP.
Logging on works fine, after logging on You see the red bar at the top of the screen "ajaxRequest declined - Referrer " in the apache2 access log you will find a 403 (denied) GET /admin/ajax.php?command=authping " 403 43
I enabled freepbx.WEBSITE (so port 80 no ssl) no change, didn’t work either. Consulting the other webpages with similair issues i did find a answer. On the page ( community-dot-freepbx-dot-org/t/can-one-disable-freepbxs-bad-referrer-check/22136/3) SkykingOH mentions a setting on the “advanced settings module” unsure which setting.
After reading and reading, I think the Ajax component is ‘hardcoded’ expecting a referrer from the localhost CentOS and or the CentOS ip and not the https freepbx.WEBSITE . In the apache2 access log you see other GET requests working flawless /admin/config.php for example.
The apache2 error log mentions : [authz_core:error] [pid 15237] [client IPfromReverseProxy:40205] AH01630: client denied by server configuration: /var/www/html/admin/index.html
Anyone has any clue? Perhaps it has something to do with the ?
Best Regards,
Tom
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Re-route calls based on sip peer availability
If you prefer using a script, that shouldn’t be hard. For example ping the office IP address every 10 seconds; if six failures in a row consider it down. Then, when it starts responding again, consider it up. Using a Call Flow Control in your routing, you could set or clear that programmatically with e.g.
asterisk -rx "database put DAYNIGHT C0 NIGHT"
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Playing Voicemail causes incoming call
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Pick up calls from queues
That is not how Queues work. What you want is Park. So that is the right solution.
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Zulu ios app
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Self Sign Certificate installation
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Search on UCP
If you have access to the Admin GUI, you can search the CDRs based on multiple fields (including dates) in the CDR Reports. Is there a reason you’re not using that?
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Re-route calls based on sip peer availability
Sterwart1, yeah that’s more along the lines of what I’m looking for. So basically run all calls through a call flow control that the script toggles on/off based on the ping responses from the location.
Now… to learn scripting with asterisk 
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Ring group spiteful, the last extension indicated does not ring ... maybe a bug?
the ring strategy is obviously about everyone
yes, the extension obviously quilted normally
the strangeness is: it does not only play the last of the list
for example if I put the inside 10 as the last one will not ring the 10 if I put the 12 will not ring the 12
The last is always the one that does not ring whatever it is
I was clear I hope!
I attach image…:
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Search on UCP
As an admin, I do use the admin GUI. But as a user, I would prefer not to give them access to freepbx (even if you can limit the feature access).
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Pick up calls from queues
Hi Tom, thanks for reply!
You are definitly right BUT i would like and hybrid between those two functionalities!
Parking is pretty good but you have to remember wich call is parked where…
it would be way better if parking was “automatized” so if you have 3 incoming calls A-B-C, if you park A then B then C when you press 1 button it will get back A, when you terminate call with A if you press the button again you will pick B and so on…
Is this achievable in any way? if not, i’ll just use the classic parking feature ^^
Thans anyway for your time!
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FreePBX + Cisco 7945 'Hello World'
Hi
I’m considering using VOIP at home, instead of PSTN, and I want to do a quick ‘hello world’ test, to get a feel of this approach. I would greatly appreciate some help, as I’m new to this space.
For this test, I’m running FreePBX on a VM (standard installation), on a MacOS. MacOS Firewall disabled, VM running on host networking (no NAT). Seems to be running OK.
I have a Cisco 7945 phone.
So far, I’ve set up a TFTP on a NAS, successfully upgraded the phone (via TFTP) to SIP45.9-4-2SR3-1S.
Below is my SEP.cnf.xml - which according to FTP logs gets served to the phone. I’ve tried to keep it minimal.
I’ve added a user account to FreePBX (as reflected by the cnf.xml).
Phone tries to register, but nothing happens (not even an error) - it keeps trying to register, and I don’t know how to proceed.
Nothing on the phone logs (except failure to update locale - which is because I don’t have the locale files on tftp).
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>cisco</sshUserId>
<sshPassword>cisco</sshPassword>
<devicePool>
<dateTimeSetting>
<dateTemplate>D/M/Y</dateTemplate>
<timeZone>Jerusalem Standard/Daylight Time</timeZone>
<ntps>
<ntp>
<name>timeserver.iix.net.il</name>
<ntpMode>Unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<sipPort>5060</sipPort>
</ports>
<processNodeName>127.0.0.1</processNodeName> <!-- I've also tried the FreePBX IP -->
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<sipProfile>
<natEnabled>false</natEnabled>
<natAddress></natAddress>
<sipProxies>
<backupProxy></backupProxy>
<backupProxyPort></backupProxyPort>
<emergencyProxy></emergencyProxy>
<emergencyProxyPort></emergencyProxyPort>
<outboundProxy></outboundProxy>
<outboundProxyPort></outboundProxyPort>
<registerWithProxy>true</registerWithProxy>
</sipProxies>
<preferredCodec>g711alaw</preferredCodec>
<phoneLabel>Y a r o n</phoneLabel>
<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>180</timerRegisterExpires>
<!-- Force short registration timeout to keep NAT connection alive -->
<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>
<sipLines>
<!-- Add lines here -->
<line button="1">
<featureID>9</featureID>
<featureLabel>Line 1</featureLabel>
<!-- Displays next to Line Number -->
<name>0001</name>
<!-- SIP username -->
<displayName>Line 2 caller ID</displayName>
<!-- Name to display on outbound caller ID -->
<contact>1</contact>
<!-- SIP username again -->
<proxy>10.1.0.34</proxy>
<!-- SIP server -->
<port>5060</port>
<!-- AUTH -->
<authName>Yaron</authName>
<!-- auth SIP username same as <name>-->
<authPassword>*****</authPassword>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messagesNumber>*97</messagesNumber>
</line>
<line button="2">
<featureID>2</featureID>
<featureLabel>Line 2</featureLabel>
<speedDialNumber>86555</speedDialNumber>
</line>
</sipLines>
<dialTemplate>dialplan.xml</dialTemplate>
</sipProfile>
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Pick up calls from queues
That’s what you use BLF’s for and do exactly what you are talking about. Queues will not work the way you think they will for this. When you put a call in the queue it will be there in it order it was put in and they will hold. There’s not way to “pull” them from the queue, the queue will want to hold them until an Agent is available to send the call to.
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