adding the ping results:
adding the ping results:
You don’t show any on your screenshots. Are you sure that there was an incoming call in the interval you are displaying?
Sorry, I See what you mean. Whenever I Call, nothing happens on wireshark neither PBX. I wonder if the gateway is not forwarding calls then?
Sorry, so lost.
Is it connected to the POTS line correctly? What does the HT Status page show for the FXO port?
Yes, POTS line is currently connected to the FXO gray port on the gateway. Please see the below for the gateway status:
Something I did notice is that when a call comes in, the FXO port does not display a “RING” status. It stays as IDLE all the time.
I’m very puzzled. Confirm that in the HT, Unconditional forward to VoIP has the correct FreePBX address (I assume 192.168.2.101). Try connecting an analog phone in place of the HT FXO port, using the same cord and confirm that it rings when you call in.
Try setting up syslog on the HT:
Syslog Server: 192.168.2.101
Syslog Level: Extra Debug
Send SIP Log: Yes
With Wireshark capturing on the PC, reboot the HT and you should see lots of syslog messages. Then, call in and report what, if anything, shows up on syslog.
I went ahead and enabled SYSLOG on the HT, and made a couple test calls. The following was recorded by wireshark:
When I hang up, thats when SYSLOG displayed the tone stopped.
Definitely calls my attention the “Port unreachable” log.
The Port Unreachable errors are not a problem. From the point of view of the Windows OS, there is no syslog server listening on UDP port 514 so it responds to each incoming syslog packet with the ICMP error. Wireshark ‘snooping’ the traffic does not count as ‘listening’ for this purpose.
I believe what is actually wrong is that the HT will talk SIP only on its WAN interface. Connect the WAN port to your regular router/firewall. Either configure it with a static 192.168.1.x address (outside the range your router’s DHCP server assigns), or let it get its address by DHCP but configure the router so it is static. The Windows host and the VM should also be connected to the router and have 192.168.1.x addresses. If you don’t have a spare router or switch port, the HT Device Mode can be set to Bridge and you can then plug the computer into the HT LAN port.
You will have to change the trunk config, HT FXO page and syslog setting to use the new 192.168.1.x addresses.
If you still have trouble but something is logged by Asterisk, paste it as described earlier. If not, paste the complete syslog data for a call: In Wireshark, select one syslog packet. Right-click and choose Follow -> UDP Stream. A new window will open showing all the syslog text. Click Save as and save it as a .txt file. Paste that at pastebin.freepbx.org and post the link here.
I found an old router I had, & as you mentioned, the IP Addresses were automatically set as described below:
Host PC: 192.168.1.2
When making a call, It doesnt seem to populate on asterisk, however the following message does come up:
Also, on Wireshark The call does come up now, please see the below:
I think it may be a port issue at this point?
This is really strange. The PBX is attempting to register to the HT. The HT is not a SIP server and that won’t work. But it is apparently a pjsip trunk, yet your config is for a chan_sip trunk and the word ‘pjsip’ (until now) didn’t even appear in this thread.
Also, the HT is attempting to register to the PBX port 5160 (which I assume is chan_sip, if you haven’t changed bind port numbers), but your HT config shows SIP Registration set to No and shows a static config for your chan_sip trunk.
OK, so you need to use pjsip or chan_sip, not both. You can use registration (from HT to PBX), or you can configure statically (the PBX trunk is manually configured with the address of the HT).
Please explain how you are trying to set this up.
Thanks for the continued support.
It might have shown that as I was making a few tests to see how it worked the best.
In your opinion, Will Chan_sip be the best route?
I followed the chan SIP Route, i See the invite to register from the HT to FreePBX, yet it seems that its being blocked, could it be fireall issues? I already listed the address as trusted:
pjsip is simplest and the way of the future, but still has some quirks that make it difficult to use both the FXS and FXO ports on the same device. If you don’t plan to use the FXS, either is fine.
If you want to use chan_sip with registration, you should have
and no port= parameter. Also, the trunk name and user= should be the same, matching SIP User ID in the HT.
Do you think the following is correct?
Looks fine, though for a reason I don’t understand, some functions require that the Trunk Name on both the General and SIP Settings tabs also be 12345.
Wohooo!! It now works! Thank you for your continued support Stewart!
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