I have posted this project before and many did not understand what they were bidding for. If you are interested, Please ...issue a hangup on sip side, i believe sip 487 Sip client on phone 1 will only register to sip server when it detects that headset jack is being used. If you can also make asterisk work as phone 1 (using chan_alsa) Please mention.
I am looking for an expert programmer in asterisk that can set all call rejection codes to 503 and Instantly reject all busy circuit calls without any delay by removing the played busy circuit massage, these mods needs to be done on 2 Freepbx Server running Asterisk 220.127.116.11
i got this error in asterisk and freepbx under centos "The number you dialed is not in service, please check your number and try again" im using twilio sip trunk, please i need soeone who fixed, everything i twilio is already done
Zoho's phonebridge plugin supports up to Asterisk 1.4 [login to view URL] With modern asterisk versions (11 and up) it's not working at all do to (apparently) java problem. And the property in the phonebridge adapter seems like it is not being set properly. With Astersik 1.8 it was working partially
I need Install Latest VICIDIAL with CENTOS 7 I need that is 100 Working.
...on will be defined by BB1, BB2, and BB3 - We have 3 cloud VPN servers based on IPsec/SSTP protocol, later on will be defined as VPN1, VPN2, and VPN3 - we also have 3 cloud Asterisk "Elastix Call Center" servers , later be defined as CC1, CC2, and CC3 - Behind the pfsense there are the agents Our typology ? Agent >> BroadBand >> Cloud VPN >> Cloud