[asterisk-users] - SIP retransmission problem

[Paolo De Michele]

Hi all,
I have a problems with asterisk
I configured a number eutelia for so I ring an internal

the file is as follows:


# sip.conf

Register: 0707732332:xxxxxxxx at voip.eutelia.it/0707732332
registertimeout20
registerattempts10

[eutelia]
type=friend
Xlanguage=it
context=from-eutelia
host=voip.eutelia.it
dtmfmode=rfc2833
directmedia=no
fromdomain=voip.eutelia.it
fromuser=0707732332
defaultuser=0707732332
insecure=invite
qualify=yes
secret=xxxxxxxxxx
disallow=all
allow=ulaw
allow=alaw

# extensions.conf

[from-eutelia]

exten => 0707732332,1,NoOp
same => n,Dial(SIP/1000,45)
same => n,Hangup

from cli:

Using SIP RTP CoS mark 5
-Performing of 0707732332from-eutelia: 1 NoOp (SIPeutelia-00000022) in new
stack
-Performing of 0707732332from-eutelia: 2 Dial (SIPeutelia-00000022,
SIP1000, 45) in new stack
Using SIP RTP CoS mark 5
-SIP1000 called
-Ringing SIP1000-00000023
-SIP1000-SIPeutelia-00000023 replied 00000022
21 Jul 23:34:23 WARNING2491: c:3641 retranspkt: chansip. retransmission
timeout reached on transmission A66023DE-D2B211E1-8D25E362-A130DAB
9195.62.226.18 for seqno 101 (critical response)-see https:
wiki.asterisk.orgwikidisplayASTSIPRetransmissions
Package timeout after 6401ms with no response
21 Jul WARNING2491: 23:34:23 c:3670 retranspkt chansip.: Hanging up call
A66023DE-D2B211E1-8D25E362-A130DAB 9195.62.226.18-no response to our
critical packages (see https: wiki.asterisk.orgwikidisplayASTSIP

the call drops after 4-5 seconds
can you tell me what do I go to edit why this doesn't happen?
I also disabled iptables in order to avoid any problem (although it seems
properly configured)

Thanks in advance
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