totsuka
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- Sep 27, 2009
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I've got PBXF3 running on a Pi (I think it is the original Pi).
Anyway, it has been running for a few years no issues with several different VOIP providers.
Yesterday I registered for a DID from DIDWW as they have a lesser used country that I needed.
I have anonymous inbound SIP calls enabled (fun to watch the hackers in the logs when I allow 5060 to the world).
I pointed DIDWW to my server, no trunk setup. I configured an inbound route with the DID number and set the destination to the extension I want it to ring at.
Made a test call, extension rings.
When I answer I can hear the caller, but the caller can't hear anything.
Asterisk was doing RTP Allocating from port range 10000 -> 15000, but this DIDWW page says they need from 10000 - 30000. So I adjusted FreePBX via the GUI to extend the listening ports up to 30000.
Restarted to be sure, same result. One way audio. Here is a snippet of the Asterisk log, it actually thinks there is no RTP activity at all.
[2017-03-23 14:10:03] VERBOSE[3319][C-00000001] app_dial.c: -- Called SIP/8526
[2017-03-23 14:10:03] VERBOSE[3319][C-00000001] app_dial.c: -- SIP/8526-00000003 is ringing
[2017-03-23 14:10:04] VERBOSE[3319][C-00000001] app_dial.c: -- SIP/8526-00000003 answered SIP/46.12.9.4-00000002
[2017-03-23 14:10:35] NOTICE[2991] chan_sip.c: Disconnecting call 'SIP/8526-00000003' for lack of RTP activity in 31 seconds
Any ideas where to look?
Anyway, it has been running for a few years no issues with several different VOIP providers.
Yesterday I registered for a DID from DIDWW as they have a lesser used country that I needed.
I have anonymous inbound SIP calls enabled (fun to watch the hackers in the logs when I allow 5060 to the world).
I pointed DIDWW to my server, no trunk setup. I configured an inbound route with the DID number and set the destination to the extension I want it to ring at.
Made a test call, extension rings.
When I answer I can hear the caller, but the caller can't hear anything.
Asterisk was doing RTP Allocating from port range 10000 -> 15000, but this DIDWW page says they need from 10000 - 30000. So I adjusted FreePBX via the GUI to extend the listening ports up to 30000.
Restarted to be sure, same result. One way audio. Here is a snippet of the Asterisk log, it actually thinks there is no RTP activity at all.
[2017-03-23 14:10:03] VERBOSE[3319][C-00000001] app_dial.c: -- Called SIP/8526
[2017-03-23 14:10:03] VERBOSE[3319][C-00000001] app_dial.c: -- SIP/8526-00000003 is ringing
[2017-03-23 14:10:04] VERBOSE[3319][C-00000001] app_dial.c: -- SIP/8526-00000003 answered SIP/46.12.9.4-00000002
[2017-03-23 14:10:35] NOTICE[2991] chan_sip.c: Disconnecting call 'SIP/8526-00000003' for lack of RTP activity in 31 seconds
Any ideas where to look?