Why SIP Trunk is Unreachable

In this guide we will tell you the reason causing sip trunk unreachable as well as extension unreachable partly.

Why Unreachable?

SIP trunk can't be registered and it shows unreachable on the web

This problem is usually caused by network problems. So please check the problem on network side first. Below are possible problems of network.

  • SIP trunk registration domain can't be parsed. Check the DNS settings in the LAN settings of PBX as well as the router settings.
  • Firewall might blokc the traffic for the IP address of SIP trunk registration server. Check the firewall relevant settings both on PBX and firewall device within the network.

 SIP trunk can be registered, but it shows unreachable

This type of problem is caused by no response of OPTIONS packet sent by PBX from the SIP trunk server.

Extension will also have similar of problem, but it show unavailable/unreachable on extension status.


There is temporary solution for this problem, but no solve the problem completely.

The way is to disable qualify which will stop sending OPTIONS packet. If issue persists, we have to check the relevant settings which could affect the OPTIONS packet traffic in the router and firewall.

  • Disable Extension Qualify:

  • Disable SIP Trunk Qualify:

SIP Qualify Mechanism

If we enable Qualify option for SIP trunk or extension, Asterisk will send a SIP OPTIONS packet periodically to check whether the device is still online or not. If the device does not answer within the configured (or default) period Asterisk considers the device off-line.

By default asterisk sends the qualify every 60 seconds. You can change this value with qualifyfreq on S series (Settings>PBX>General>SIP>Qualify Frequency). The value of Qualify represents the timeout after a packet is sent before we consider the peer to be unreachable. If the packet is not responded within 1 second, asterisk will keep trying until 7 packets have failed. At this point, asterisk won't try again until the next 60 cycle period completes. If a packet is lost, which can easily happen with UDP, there are 7 more packets which are transmitted. Additionally asterisk will keep trying every 60 seconds. So even if all 7 packets are lost, asterisk tries again at the next 60 second cycle. 

Reference: http://www.voip-info.org/wiki/view/Asterisk+sip+qualify

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