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Change Asterisk Server Config to not need ACK on SIP Trunk RTP traffic
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IT & Programming > Technical Support

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  • Posted: Sun, Aug 28, 2011
  • Time Left: Closed
  • Location: Anywhere
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  • Start: Immediately
  • Budget: Less than $500
  • Fixed Price Job
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A working Asterisk server has been in place for about 5 years. For no apparent reason, the asterisk server has begun to drop calls made to the Asterisk server after exactly 20 seonds.

We have done some packet sniffing, and have found that our site router is dropping some ACK packets coming back from the ITSP going to the Asterisk server. These packets confirm that the RTP traffic is flowing. When these RTP packets are not recieved within 20 seconds, the Asterisk server drops the call.

When the call is dropped, the calling party does not get terminated, the local LAN handset connected to the Asterisk server just says 'call ended' and hangs up.

We did some packet sniffing and capture diagnosis (Asterisk server is 172.16.171.20, site firewall/router is 172.16.171.1) with our ITSP and they analysed it and said:
"From the capture, I can see the call was made from 61987659058 to 0987659058. Not sure why
the capture shows the line calling itself. Do you have any idea?
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Desired Skills
VoIP
Job ID: 25878153
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