Hello everyone
I have a Kazoo cluster setup in a VPC on AWS and experiencing a call drop issue.
I can make an outbound call from a client registered on Kazoo (a deskphone or softphone). I have full audio both ways, but the call is dropped at 30 seconds.
The freeswitch logs say:
Via: SIP/2.0/UDP 54.148.57.6:11000;rport;branch=z9hG4bKDtpjFF2FZQgZg
Route: <sip:10.20.30.5;lr=on;ftag=as15b0c5a8>
Max-Forwards: 70
CSeq: 74186631 BYE
Contact: <sip:2063130566@54.148.57.6:11000;transport=udp> User-Agent: 2600hz
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
Reason: SIP;cause=408;text="ACK Timeout"
Content-Length: 0
So it looks like its getting an ACK timeout.
My profile status
Name sipinterface_1
Domain Name N/A
Auto-NAT false
DBName sofia_reg_sipinterface_1
Pres Hosts
Dialplan XML
Context context_2
Challenge Realm auto_from
RTP-IP 10.20.30.12
Ext-RTP-IP 54.148.57.6
SIP-IP 10.20.30.12
Ext-SIP-IP 54.148.57.6
BIND-URL sip:mod_...@54.148.57.6:11000;maddr=10.20.30.12;transport=udp,tcp
HOLD-MUSIC local_stream://default
OUTBOUND-PROXY N/A
CODECS IN H263,OPUS,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,G729,GSM,Speex
CODECS OUT H263,OPUS,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,G729,GSM,Speex
TEL-EVENT 101
DTMF-MODE rfc2833
CNG 13
SESSION-TO 0
MAX-DIALOG 1000
NOMEDIA false
LATE-NEG true
PROXY-MEDIA false
ZRTP-PASSTHRU false
AGGRESSIVENAT false
CALLS-IN 60
FAILED-CALLS-IN 25
CALLS-OUT 22
FAILED-CALLS-OUT 5
REGISTRATIONS 0
The dbtext/dispatcher has the right IP. Both sides of the call get audio. At the 30 second mark, the call is dropped.