Our SIP Trunk provider is sending an invite but the freepbx replying a “100 Trying”
After 32 seconds the call was cut. This is only happening on inbound call. Outbound call is working fine.
Our FreePBX are connected directly to our provider’s router so there is no NAT or Firewall in between.
Our Trunk Provider saying that they are not receiving a 180 ring response.
In Asterisk SIP Settings, confirm that External Address and Local Networks are correctly set.
Based on what you have told us so far, Local Networks should be 10.0.0.0 / 8
However, if you have a second NIC with a different private address for your phones and/or for connecting to the internet, you should have additional entries for those.
If you change any of these settings, you must restart (not just reload) Asterisk.
If you still have trouble, paste the complete SIP trace (not just the incoming INVITE) for a failing call at pastebin.freepbx.org and post the link here. Please do not post textual data as an image.
Hi Stewart1, sorry for posting the textual data as an image.
My new account is not allowing me to paste links and I’m not aware yet that the forum has its own pastebin.
Your OP says “Not Sending ACK”. However, the SIP trace shows ACK is being received by pjsip but apparently not recognized, so it retransmits the 200 OK, the gateway sends another ACK, etc., until the call is disconnected after 32 seconds for lack of response.
Unfortunately, I don’t know why pjsip didn’t ‘like’ the ACK. On (for example) lines 177 and 178 there are two Content-Length headers. This is apparently a bug on the provider’s side and is not valid SIP, but it is not obvious why a robust SIP stack should reject it, especially because the incoming INVITE has both the Content-Length and Content-Type headers duplicated, and pjsip accepted that just fine.
Further confusing the issue, I made a test call to Truelogic and kept the call up for well over 32 seconds by repeatedly entering invalid extension numbers, and the call did not drop. Have you fixed the problem, was my call (from +1 775 xxx xxxx) answered by another system, or is the trouble intermittent?
If you are still having problems, at the Asterisk command prompt type pjsip set logger on
make a failing incoming call, paste the Asterisk log (not the console log) for the call and post the link. The Asterisk log should contain both the dial plan flow and the SIP trace intermixed and with luck there will be an error from pjsip about a problem with the ACK.
Thank you for calling our number that’s lead me to recheck our CDR.
I noticed that all incoming call with caller id that start with + was hang up after 32 seconds.
All caller id that has no + kept the call over 32 seconds like the one you did.
I confirmed it by calling from US and skype number.
The + sign on caller id only started when our trunk provider replaced their router connected to our pbx.
I already informed them to remove the + on all caller id of incoming call.