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Substitute CallerId "+" with "00"

@Paolodic wrote: Hi,is there an easy way to substitute any incoming CallerId beginning with + with 00? It would be helpful for telephones which only cut the + because they are not able to handle.I've...

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Conference Call and VAD

@FrankRadio wrote: Can I setup into freepbx conference configuration the VAD (Voice audio detection) ON? I don't want that the conference streams audio until someone talks, I mean I don't want any...

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SIP peer lagged and then unreachable intermittently

@jhelou wrote: I have a Sangoma Vega50 FXO gateway that is trunked into a current version freepbx VM running on a windows 2016 hyper-v. This VM is a restore of a freepbx physical server that i was...

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Couple of Errors on new Distro Install and Restore

@hydro3 wrote: Hello, I am trying to restore to a new machine. The original was system was/is a distro install on FreePBX13.0.191.19. I did a full backup and then moved the file onto a new system that...

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Deadair when using AMI originate between two external destinations

@DoctorB wrote: Hey all, I have a piece of PHP script that telnets into Asterisk AMI (FreePBX 13.14) on port 5038 and originates a call between two given external numbers A and B. I have done a few...

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Mail Delivery Subsystem - Adress Not Found - asterisk@mydomain

@Geloso wrote: Hi there. I have a freepbx hosted in a cloud service (specifically for freepbx)I am in the testing face, so it's not in production. But testing is going well. I have the system admin...

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PJSip 'Allow Guests' - what does this do exactly?

@dickson wrote: Got 100 or so Avaya 1100 series SIP phones pointing to an old Asterisk box. Want to point them to a new asterisk box and use PJSIP. Got the test batch of phones working, but in the log...

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Can't see fax option in UCP

@Nucleus wrote: Using FreePBX 13.0.190.19 In User Manager, I created a user linked with an extension. In User Manager, in the FAX tab, it is enabled and properly configured When I log in with that...

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How to manage USER/PASS registration on FreePBX

@its wrote: Hello, i'm looking for ways to allow PBX users system to force registration of the device in use (i.e. Yealink phone) via script or sending user/pass via web.This way, each user can use...

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MWI hitting phones and making notifications over and over

@bigmillz wrote: Common complaint from users that seems to have started recently. User A has a voicemail he hasn't listened to yet. Clears the notification on phone and goes back to work. An...

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SIP trunk etisalat UAE

@waseem50 wrote: Hello all,my telecome provider etisalat / UAE has installed a sip trunk in my office. i use freepbx .the trunk works perfectly with xlite however i cant seem to be able to register...

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Dial Plan to manage which extenstion can be dialed and which not

@Geloso wrote: Hi there. I have 2 companies in a freepbx. Company 1 (C1) (with extensions range from 100 to 199) can dial only extension in the range of 100 to 199 Company 2 (C2) (with extensions...

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FreePBX sending OK 200 every second

@abdel33 wrote: In my newly installed Freepbx system, the freepbx is sending 200 OK message to one of our my trunks every second? I even restarted the asterisk application to see if it stops but it is...

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2 Servers - Missed call notifications

@dpclark wrote: HI, We have 2 Servers with a IAX trunk between them - both running 13.0.190.19 We can make calls between the two quite easily and the 2nd server uses the 1st to make outgoing calls as...

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Convert a SIP Extension to a Virtual Extension

@Adamski wrote: I have a user that has semi-retired.He'll no longer need a desk phone, just access to voicemail.He doesn't want calls to follow him. Just hit voicemail and he'll check the voicemail...

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Trunk Not Registering

@ameeno wrote: Hello there. I Have a sip trunk that will not allow my registration string. this server needs a reply from my pbx before it allows my pbx's ip through, apparently my pbx is not...

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Sipstation Module Errors

@kbocek wrote: I'm trying to experiment with the free trial of the Sipstation service. I installed the module and then tried to run it. But it gave some errors when I tried to enter my email. The...

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System admin update error

@assos40 wrote: Using Freepbx 10.13.66-19.While selecting 'Updates' on System Admin i getfopen(http://upgrades.schmoozecom.net/updates.php): failed to open stream: HTTP request failed!System Admin is...

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Asterisk Chan_Unistim

@nortelvoip wrote: Hello, I'm on FreePBX version 13 and Asterisk version 13.14. I was on Elastix 2.5 but recently decided to move to FreePBX. On Elastix my Asterisk system had chan_unistim...

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How to make a disconnected extension ring 3 times before voicemail?

@cmariotti wrote: I have a ring group that calls a specific extension that does not have a phone hooked up... It's working great, but we need it to RING THREE TIMES BEFORE going Voicemail. Any idea of...

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