Hello everybody, this is my first post, although i have been using 3cx for more than 6 months now
I am using Patton 4638 5port BRI Gateway with Linksys (921 & 961) IP SIP Phones
My 3CX version is 5.0.3752.0
I have setup the Patton gateway through the automated 3cx process and the saved configuration file, and all lines register fine
When i use the standard configuration, the lines work correctly both in and out, but when i try to transfer a line from the SIP Phone that answered the call, the caller will not hear my on-hold music, but a busy-like tone instead (which i suspect comes from the Patton device). This only applies to calls with the PSTN, as internal hold/transfer plays the default wav file with no problem
If i try to disable "Supports Re-Invite" on the gateway, then the caller can hear the wav, but once the transfer is complete, the caller is on silence, the calls are disconnected, and the call recipient sees the caller on hold.
There is no error indication on the log files, just that the call ends as soon as the caller (isdn) and the recipient (sip) both join in SIP
I will try to update to the newer version, as there seem to be some issues although from a test server that i set it up on, i experienced the same issues, without working too much on it tho.
I am using Patton 4638 5port BRI Gateway with Linksys (921 & 961) IP SIP Phones
My 3CX version is 5.0.3752.0
I have setup the Patton gateway through the automated 3cx process and the saved configuration file, and all lines register fine
When i use the standard configuration, the lines work correctly both in and out, but when i try to transfer a line from the SIP Phone that answered the call, the caller will not hear my on-hold music, but a busy-like tone instead (which i suspect comes from the Patton device). This only applies to calls with the PSTN, as internal hold/transfer plays the default wav file with no problem
If i try to disable "Supports Re-Invite" on the gateway, then the caller can hear the wav, but once the transfer is complete, the caller is on silence, the calls are disconnected, and the call recipient sees the caller on hold.
There is no error indication on the log files, just that the call ends as soon as the caller (isdn) and the recipient (sip) both join in SIP
I will try to update to the newer version, as there seem to be some issues although from a test server that i set it up on, i experienced the same issues, without working too much on it tho.
