Call hangs up due to retransmission time-out

Hi everyone,

I’m trying to get get a call working from a softphone with an external sip account (express talk, sisp2sip.info) to sipml5 with a sip account that is registered to my server, so far I can get the call connecting and audio flowing but, on calls to sipml, the call hangs up after 32 seconds (the call is fine from sipml to express talk though) I’ve checked the traces and it seems to be a retransmission error however after goggling the issue I’ve been unable to fix it my self my config files are:

sip.conf:

[code][general]
context=unauthenticated
allowguest=yes
udpbindaddr=0.0.0.0:5060
tcpenable=0
realm=serverIPAddress
transport=udp,ws,wss
;codecs
disallow=all
allow=ulaw
allow=alaw
allow=gsm
nat=force_rport,comedia

webRTCcnct
type=friend
context=from-internal
host=dynamic
icesupport=yes
secret=****
encryption=yes
avpf=yes

1060
defaultuser=1060
dial=SIP/1060

1061
defaultuser=1061
dial=SIP/1061
[/code]

Extensions.conf

[code][globals]
FIRST=SIP/1060
SECOND=SIP/1061
[unauthenticated]
exten => 1060,1,Dial(${FIRST},20)

exten => 1061,1,Dial(${SECOND},20)

exten => 200,1,Answer()
same => n,Playback(hello-world)
;same => n,Hangup()

[from-internal]
exten => 1060,1,Dial(${FIRST},20)

exten => 1061,1,Dial(${SECOND},20)

exten => 200,1,Answer()
same => n,Playback(hello-world)
same => n,Hangup()

[/code]

http.conf:

[general] enabled=yes bindaddr=0.0.0.0 bindport=8088

My trace from sipml5

My trace from asterisk with sip debug on

Does anyone have any Idea how to solve this? any help would be greatly appreciated

You are not configured to access systems outside your NAT. You need to specify the external address or a STUN server.

In my rtp.conf file I’ve specified a stun server(forgot I had changed that so I’ll post it now)

[general]
rtpstart=10000
rtpend=20000
icesupport=true
stunaddr=stun.l.google.com:19302

I thought stun was an alternative to externaddr and externhost, but it looks like you may need one of those. Also check localnet(s).