In this guide, we list several reasons that cause SIP trunk unreachable or extension unreachable on S-Series VoIP PBX and provide the corresponding solutions.
Why SIP Trunk is unreachable?
Firstly, the problem is usually caused by network problems. So check the problem on network side first. Below is the related settings:
- 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 block 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.
Secondly, it can be caused by no response of OPTIONS packet from the SIP trunk server which the packet is sent by PBX.
Actually it is the problem OPTIONS packet conversation failure due to the router / provider.
- OPTIONS packet is redirected by the router.
Normally if PBX sends a OPTIONS packet out, it should receive an response no matter 200 OK other response. But if the OPTIONS packet has been modified by the router, the other side could not receive the OPTIONS packet or it could not reply the response to the correct source IP address. You could refer this case:
- The provider doesn't reply the OPTIONS packet sent by PBX.
There is a temporary solution to this problem, but it will not solve the problem completely.
The way is to disable the "Qualify" option on the SIP Trunk advanced settings or extension advanced settings, which will stop ending OPTIONS packet. If the issue persists, we'll 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:
alt="Edit SIP trunk on PBX"
- If above solution doesn't work. It could be that problem that the provider requires PBX to send OPTIONS to prove it is still alive. You can refer the cases in the following
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 will consider the device off-line.
By default, Asterisk sends a SIP OPTIONS packet every 60 seconds. You can change this value with Qualify Frequency settings on S-Series VoIP PBX (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-second 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.