We are experiencing a weird problem with our PBX.
The internal calls are working just fine.
But the external are not, two out of five calls go through with both ways audio, rest with only one way.
Also when we make a call that happens to be the one way audio we get this in the log:
[2014-12-16 13:36:40] SECURITY: res_security_log.c:134 security_event_cb: SecurityEvent=“ChallengeSent”,EventTV=“1418737000-378439”,Severity=“Informational”,Service=“SIP”,EventVersion=“1”,AccountID=“sip:firstname.lastname@example.org”,SessionID=“0x14c4bf28”,LocalAddress="IPV4/UDP/81.../5060",RemoteAddress=“IPV4/UDP/220.127.116.11/5060”,Challenge=“42dde94e”
[2014-12-16 13:36:47] NOTICE[C-00000026]: app_dial.c:958 do_forward: Not accepting call completion offers from call-forward recipient Local/5**@from-internal-00000011;1
[2014-12-16 13:36:47] WARNING[C-00000026]: func_presencestate.c:132 presence_read: PRESENCE_STATE unknown
The system is up to date.
I recently went through a nightmare involving this and there’s a lot to deal with! Let me see if I can help you understand what’s happening (I’m not a guru like some of the folks here, but I’m learning )
Before I start, I’d like to let you know that exposing any of these IP addresses is probably a bad idea. I usually put some indicator of what the IP is for, such as [serverip] or [phoneip] instead.
One-way audio isn’t a FreePBX problem, it’s a networking problem. The RTP traffic is not getting through, which can be at any number of points. The Settings > Asterisk SIP Settings page contains the NAT settings for your server, and the router on the endpoint should be allowing the traffic through. Additionally, there may be a ‘SIP ALG’ setting on that router; these settings should usually be turned off as they typically do more harm than good. With that said, if it is off, you could try switching it on as a troubleshooting step. That SIP Settings page will be important and make sure the IP info in there is correct.
The logs here don’t appear to be related, these are all pretty standard messages. The call forward one could be problematic, but it’s likely unrelated to your issue.
You should also make a habit of posting version numbers; Asterisk and FreePBX version info is helpful for troubleshoot ing. For example, after some trial/error/reading, our production system is on Asterisk 11.14.2 running FreePBX 12.0.19 on CentOS 6.6. All of these things help narrow down the source of an issue.