Outbound Call Answered by s [ivr-1]

I am currently migrating some numbers from FreePBX to another PBX. I wanted to leave the 4 digit ext and just forward them to a misc destination. When forwarding, the call rings once and then is answered by the FreePBX. Below is the CDR of the “failed call”. I removed the DID, but something in the FreePBX still thinks this number lives here. This same issue happens whether its to the 10 digit or the 4 digit ext. The 10 digit is where the problem is. I jsut cant find anywhere else its built. I wasnt sure if it was cached somewhere or I am running into another issue.

	BackGround	s [ivr-1]	ANSWERED	00:04
                  Channel: Destination Context: ivr-1

It’s common to have an Outbound Route that matches the PSTN number range used by your PBX and sends the call via a loopback trunk. This avoids using a provider trunk and its associated cost, latency, etc. If that’s your situation, modify the dial pattern so it no longer matches numbers that have been moved to the new PBX.

Otherwise, post some details: How is the forwarding done? If you call the forwarded-to number from a FreePBX extension, does that work correctly? Post the Asterisk log for a failing call.

Stewart1,

I see two legs on the call, which are below. When dialing out I hear one ring then I see the call connect and its dead air. Same result on a call that is forwarded. The loopback piece makes sense, but I dont see anything in the outbound routes, I also tried to build an outbound route just for the single DID to correct. I should also note, the forwarding doesnt really make a difference, same result when the ext or the 10 digit is dialed.

FreePBX version 13.0.192.19

[[email protected] ~]# grep 1545402563.96336 /var/log/asterisk/full*
/var/log/asterisk/full:[2018-12-21 08:29:23] VERBOSE[23396][C-000053ba] pbx.c: Executing [[email protected]:1] Set(“SIP/6854-0000a252”, “TOUCH_MONITOR=1545402563.96336”) in new stack
/var/log/asterisk/full:[2018-12-21 08:29:23] VERBOSE[23396][C-000053ba] pbx.c: Executing [[email protected]:18] Set(“SIP/6854-0000a252”, “__CALLFILENAME=out-3196654777-6854-20181221-082923-1545402563.96336”) in new stack
/var/log/asterisk/full:[2018-12-21 08:29:23] VERBOSE[23396][C-000053ba] pbx.c: Executing [[email protected]:19] MixMonitor(“SIP/6854-0000a252”, “2018/12/21/out-3196654777-6854-20181221-082923-1545402563.96336.wav,abi(LOCAL_MIXMON_ID),”) in new stack
/var/log/asterisk/full:[2018-12-21 08:29:23] VERBOSE[23396][C-000053ba] pbx.c: Executing [[email protected]:23] Set(“SIP/6854-0000a252”, “CDR(recordingfile)=out-3196654777-6854-20181221-082923-1545402563.96336.wav”) in new stack
/var/log/asterisk/full:[2018-12-21 08:29:28] VERBOSE[23396][C-000053ba] pbx.c: Executing [[email protected]:4] NoOp(“SIP/6854-0000a252”, “MASTER CHANNEL: 1545402563.96336 = 1545402563.96336”) in new stack

[[email protected] ~]# grep 1545402565.96338 /var/log/asterisk/full*
/var/log/asterisk/full:[2018-12-21 08:29:25] VERBOSE[23401][C-000053bb] pbx.c: Executing [[email protected]:22] Set(“SIP/intelligentcontacts-a1-0000a254”, “__CRM_LINKEDID=1545402565.96338”) in new stack
/var/log/asterisk/full:[2018-12-21 08:29:29] VERBOSE[23401][C-000053bb] pbx.c: Executing [[email protected]:4] NoOp(“SIP/intelligentcontacts-a1-0000a254”, “MASTER CHANNEL: 1545402565.96338 = 1545402565.96338”) in new stack

The log snippets don’t seem relevant to the problem. Post everything logged for the failing call. Paste it at http://pastebin.freepbx.org/ and post the link here. Also include info on what you dialed (if not just the 10-digit number) and what you expected to happen (send call over PSTN, send via special trunk to new PBX, etc.)

I agree, the issue is that I am just trying to dial a 10 digit number that used to exist on the PBX. It now exists externally and I am unable to route it to an external trunk. I dont see any routes inbound for the DID or the entire block of DID’s. I have 2 numbers that were removed, one of them works and one doesnt. The paste is the 2 logs of the calls. The below is what I can see is different, but I have no idea where this is being done or how to change.

[[email protected]:1]

https://pastebin.freepbx.org/view/b80fcc3b

This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.