Incoming calls dropped after 30 seconds

Hi all, i have a problem with incoming calls on a trunk: after 30 seconds communication is dropped

[quote]WARNING[4833]: chan_sip.c:4174 retrans_pkt: Retransmission timeout reached on transmission 43372B08-5F9A11E5-9632E4F6-6135C71A@83.211.2.218 for seqno 101 (Critical Response) – See wiki.asterisk.org/wiki/display/ … nsmissions
Packet timed out after 32001ms with no response
[2015-09-21 15:20:23] WARNING[4833]: chan_sip.c:4203 retrans_pkt: Hanging up call 43372B08-5F9A11E5-9632E4F6-6135C71A@83.211.2.218 - no reply to our critical packet (see wiki.asterisk.org/wiki/display/ … nsmissions).
[/quote]

i have this configuration on freepbx trunk:

PEER:

[quote]context=from-sip-external
host=voip.eutelia.it
insecure=invite
secret=mysecret
type=friend
username=myusername
user=myusername
fromuser=myusername[/quote]

USER:

[quote]context=from-sip-external
host=voip.eutelia.it
insecure=very
secret=mypassword
type=friend
username=myusername
user=myusername
fromuser=myusername[/quote]

REGISTRATION STRING:

Here’s my sip_general_additional.conf (generated by freepbx):

[quote];--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. ;
;--------------------------------------------------------------------------------;
; For information on adding additional paramaters to this file, please visit the ;
; FreePBX.org wiki page, or ask on IRC. This file was created by the new FreePBX ;
; BMO - Big Module Object. Any similarity in naming with BMO from Adventure Time ;
; is totally deliberate. ;
;--------------------------------------------------------------------------------;
faxdetect=no
vmexten=*97
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
useragent=FPBX-12.0.76.1(11.6.0)
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g729
allow=h264
allow=mpeg4
allow=h263p
allow=h261
allow=h263
callevents=no
rtpstart=10000
rtpend=20000
jbenable=no
allowguest=yes
srvlookup=no
defaultexpiry=120
minexpiry=60
maxexpiry=3600
registerattempts=0
registertimeout=20
g726nonstandard=no
videosupport=yes
maxcallbitrate=384
canreinvite=no
rtptimeout=15
rtpholdtimeout=300
rtpkeepalive=1
checkmwi=10
notifyringing=yes
notifyhold=yes
nat=yes
externip=mypublicip
ALLOW_SIP_ANON=no
localnet=192.168.1.0/24[/quote]

The peer is not receiving the final response, or the peer is not sending ACK, or the ACK is not reaching Asterisk.

Although you have a typical cook book configuration, with deprecated options and inadvisable ones, nothing would specifically cause your problem. You should look into NAT and firewall settings (NB Asterisk has not been configured to work properly behind NAT in your configuration).

Also, do not assume that everyone uses SIP, or even more specifically, everyone uses the older, chan_sip, driver. This information had to be gleaned from the logs. Also don’t assume everyone uses the same version of Asterisk. The fact that this works at all suggest you are using an obsolete version.

You know what is really strange? That same account on a soft phone on the same network perfectly works, so firewall seems not to be involved.