No outbound audio

Hello,

Looking for ideas on my issue of no outbound audio. They system has been up and running for a year or so but today there is a lack of outbound audio. After normal business hours the calls get forwarded to an after hours service (real people that take messages). Since the system does not detect outbound audio it winds up dropping the call in about 10 secs.

I’ve just started reading through some of the forum links but nothing is jumping out as a resolution. I’m hoping someone smart will say ‘oh, you need to check X first’.

There is a mix of analog and SipStation lines. The SIP lines would generate an immediate ‘your call cannot be connected…’ message. So I killed those outbound lines as a test (side note - not a big fan of SIP station). Now I’m going out through the analog lines and get the 10 second drop.

Calling into the office I get the normal IVR, can move around and choose extensions, and do not get dropped. If I call in from my cell and choose my remote extension (a VPN exists between the office and my home where I have an office extension) I verify that incoming audio is received but never hear the remote/outbound audio on my cell.

The setup is FreePBX 2.9.0.4 on top of Asterisk 1.6.2.11

I’ve read some posts about NAT issues causing one way audio but I don’t think that applies since I’m using analog lines right now.

Now I’m just floundering. When I call the back line (this goes to a ring group and there is someone at the office right now) audio work fine both ways. Argh.

Just double checked the main line and could also get through with no audio issues to the same person onsite.

Added my cell as a Misc Destination, and then as an option on the afterhours IVR, get an ‘all circuits are busy now’ when I try the option. There are 8 analog lines and 4 SIP lines available. The 4 SIP lines are set for office only outbound. The Outbound Router is supposed to use up the analog first and then the SIP lines. At one point I did have SIP lines be the first to dial out afterhours due to the volume drop I get when someone dials in and is forwarded out over an analog line.

Time to start checking the SIP setup I guess.