Running Asterisk 13 with FreePBX13 on Ubuntu 14.04LTS on internal LAN (no NAT). All updates have been installed up to today.
I have 2 extension numbers/users… 1488 and 1573
If I log into UCP on 2 PC’s, one as 1488 and one as 1573 both extensions show as ‘Available’ and both are registered in Asterisk (991488 and 991573). I get a green telephone on both UCP screens.
I can call from one UCP client to the other and it rings. I click the ‘Answer’ button and the receiving extension goes dead while the dialling extension gets diverted to VM.
I get the following in the Asterisk log…
[2015-11-18 10:45:31] VERBOSE[C-00000100] app_dial.c: Called SIP/991573
[2015-11-18 10:45:31] VERBOSE[C-00000100] app_dial.c: Connected line update to SIP/991488-0000016d prevented.
[2015-11-18 10:45:31] VERBOSE[C-00000100] app_dial.c: SIP/991573-0000016e is ringing
[2015-11-18 10:45:34] VERBOSE[C-00000100] app_dial.c: Everyone is busy/congested at this time (2:0/0/2)
If I use SIP deskphones instead of UCP everything works perfectly.
If I use SIP softphones instead of UCP everything works perfectly.
If I use a mixture of SIP soft/desk phones everything works perfectly.
If I dial from UCP TO a SIP phone (soft or desk) everything works perfectly.
If I dial from a SIP phone to UCP I get the same problem.
The issue appears to relate to calls that are incoming to a UCP agent only.
Codecs can’t be an issue as calls from UCP to SIP work.
UCP to UCP calls are point-to-point so, once the session is established it would be a direct IP stream between the 2 browsers. I’m guessing the issue is here??? Possibly an SRTP/DTLS connection issue? Not sure why though.
Anyone got any ideas?
Thanks in advance