– SIP Retransmission Problem

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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@voip.eutelia.it/0707732332

type=fiend Xlanguage=it context=fom-eutelia host=voip.eutelia.it dtmfmode=rfc2833
directmedia=no fromdomain=voip.eutelia.it fromuser07732332
insecure=invite qualify=yes secret=xxxxxxxxxx disallow=al allow=ulaw allow=aaw

# extensions.conf


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

One thought on - – SIP Retransmission Problem

  • Hi Paolo,

    I had yesterday a similar problem and it was caused by a misconfigured IP address in extensions.conf that I forgot to update after changing some IP addresses in my network.

    Check the network connectivity between you Asterisk host and “1000”. Double check that the IP address is correct. Use tcpdump to see what’s going on the wires.

    Good luck!

    Jorge Martínez López http://www.jorgeml.net