i have some strange problem. Whenever i call someone, everything works fine. Rining works, talking works, but when the person i call, put me on hold to forward me to someone else, the call stops. (I’ve tested calling several companies)
Well… I’m not a pro. This problem makes me crazy.
My configuration and setup are pretty basic. (most are the default config)
I have a normal home-network. Simple LAN. FreePBX(14 current version) is running on a small Linux-server.
Very basic extension config. I’ve set the extensionnumber and a password. (all phones works).
Basic outgoing route.
I’m using CHAN_PJSIP on port 5060.
I have a plug&play router. (my router registeres the trunk to the provider).
FreePBX registeres to the router. (password and IP).
The provider doesnt allow freepbx to register direct.
What are the first steps to figure out the problem? Seeing Logs? I’ve tried. Was too much and i think the wrong log.
You may want to watch the Asterisk command line during the call, and watch the time during the hold/transfer takes place. Then go into the the web interface and select Reports -> Asterisk Logfiles, paste the section which has the concerned call to FreePBX Pastebin, and post the link here
I will try it on monday.
" asterisk -rvvvvvvvvvvvvvvvvvv "
or do i need to:
core set verbose 100
core set debug 100
pjsip set logger on
but i think that was to much.
You do not need all that. We just want to watch the general activity of the PBX to see what happening with the call.
‘asterisk -rvvvv’ is just fine
Ah i got it^^
Heres the Link to my Log:
At the end you can clearly see the attendedtransfer-rec-restart.php call from the “hold” and then the “Hangup”.
But i don’t see any cause why the call hung up.
Maybe they are doing the procedure incorrectly? They should be pressing the transfer button, not the hold
i dont think so, we call big international companies and everytime the hold leads to a hungup.
This may be tricky to fix. I expect what is happening when the far end places your call on hold, is that the sip signalling for the hold is being transmitted through your SIP trunk to your PBX. If you can confirm that is the problem using packet captures (or sngrep), you need to figure out how to suppress that. Hold is commonly done with a SIP re(INVITE), so you might look at the various SIP settings that control reinvites. Your SIP provider may be able to assist.
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