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Spiceworks Forums » Asterisk PBX
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Recent discussions in the Spiceworks Community group Asterisk PBX.
Spiceworks Forums » Asterisk PBX
5M ago
I am looking new sip trunk providers for vitalpbx ..read more
Spiceworks Forums » Asterisk PBX
7M ago
Any ideas how to fix this
VitalPbx Intrusion Detection Stopped | Wont start I start it but it then stops ..read more
Spiceworks Forums » Asterisk PBX
7M ago
does anyone know how to setup incoming caller ID with vitalpbx for sip trunks we are using twillio.
We just switched from freepbx that had the caller ID incoming vitalpbx does not any ideas ..read more
Spiceworks Forums » Asterisk PBX
9M ago
A FreePBX v13 is sending notifications when users are added and when backups complete but no voicemail notification or messages. Problem determination isn't going well. Can I eliminate the postfix email system as a problem source given the new user and backup emails getting sent? Any suggestions on what I should focus on or what to check configuration-wise in the voicemail system would be welcome; I've been all over the this and nothing is popping.   ..read more
Spiceworks Forums » Asterisk PBX
9M ago
We are having issues with adding the remote directory from Contact Manager to Yealink phones
We have done the following:
I have followed through the below guidance:
https://community.spiceworks.com/topic/2280042-freepbx-moved-to-new-host-now-yealink-phones-can-t-do...
and through the following one:
https://mangolassi.it/topic/18647/freepbx-contact-manager-to-yealink-address-book Opens a new window
have opened the TFTP port
UTF-8 encoding for php and xml files
tried the following remote URLs:
https://ipaddress/cm_to_yl_ab.php Opens a new window
http://ipaddress/cm_to_yl_ab.php Opens a new windo ..read more
Spiceworks Forums » Asterisk PBX
10M ago
hi all,
lets say theres a country that cant access your voip server, would it make a difference if i enable SIP TCP aswell as SIP UDP under chan_sip?
ive changed the port, instead of 5160 its 51160 and that works as i can connect to my voip server on my mobile on mobile data
thanks,
rob ..read more
Spiceworks Forums » Asterisk PBX
10M ago
Hi All!
We are using WebRTC to integrate phone functions into a custom coded CRM. Our PBX platform is a self hosted FreePBX v15 box, which has been working flawlessly using SIP extensions for several years. We have made all the normal changes needed for WebRTC.
Everything about WebRTC works great except one detail. If the user calls a number, and it rings for more than 20 seconds before being answered, there is a roughly 10 second delay before the audio is connected.
We have tried spinning up a test PBX in a different datacenter, used a public STUN server, using a self hosted STUN server, and ..read more
Spiceworks Forums » Asterisk PBX
10M ago
Im using NEC SL2100 and AVAYA IP Office, Im trying to register the SIP Extension of avaya as SIP trunk of NEC SL2100.
I Try to resgister the IP address of IP Phone of avaya to SL2100 by siptrunk configuration, then configure the VOIP address of NEC to avaya.
And i try to test a call from avaya to NEC is good but NEC to Avaya is always disconnected.
Thank you in advance for your help.  ..read more
Spiceworks Forums » Asterisk PBX
10M ago
Hello everyone,
Hope u all doing great,
i need your help guys, i have noises when calling or receiving calls, the materiels i'm using are: huawei optix rtn 905, line card orange, dell server with centos and elastix installed on it, using linphone for our calls and jabra headphones, in our huawei rtn there is a red light on SRV i've access to it using WebLCT and the issue of this was on EG6 it's a "critical alarm" so i tried on other huawei not fonctionel and i fond that the issue caused by port-3 was enabled, cause i'm using only two port one for line card and second for our server, when disab ..read more
Spiceworks Forums » Asterisk PBX
10M ago
https://www.freepbx.org/oem-program/ Opens a new window
Does anyone know the cost of the freepbx OEM ..read more