Hello, we are starting to use FreePBX 13 for our office. We have an external trunk that works well in dialout and dialin scenarios for the person placing or taking the call. So I suppose we do not have the usual NAT problem situation.
We also can do internal-to internal calls with two-way audio so I also assume our local network settings must be correct. However, after doing an unattended transfer on an incoming call we cannot hear the voice of the internal person finally taking the call. We use Grandstream 2130 and 216 telephones; the issue is there regardless whether I use the transfer button of the phone, the ##xx sequence or the *2 sequence (which in principle all work, i.e. I see no error messages in the log files).
We use chan_sip for internal connections and pjsip for the external trunk.
The iptables firewall on the freepbx system is disabled and allows all packets.
In the generated asterisk config for the extension (sip_additional.conf) I see
deny=0.0.0.0/0.0.0.0
secret=(…)
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
session-timers=accept
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=udp,tcp,tls
avpf=no
force_avp=no
icesupport=no
encryption=no
namedcallgroup=
namedpickupgroup=
dial=SIP/27
mailbox=27@device
permit=0.0.0.0/0.0.0.0
callerid=(***)
callcounter=yes
faxdetect=no
cc_monitor_policy=generic
which looks OK to me.
Also we have
pjsip.endpoint.conf:t38_udptl_nat=no
and
sip_general_additional.conf:nat=no
The phones and the freepbx are in the 192.168.12.0/24 network and these are the settings for the network:
pjsip.transports.conf:local_net=192.168.12.0/24
sip_general_additional.conf:localnet=192.168.12.0/24
We tried changing the internal driver to chan_pjsip also but then we have “no-way audio”, i.e. no side hears anything.
What are we doing wrong?