Call recording issue


Been having some issues with call recording. Any help or suggestions or even point to the right direction is greatly appreciated. The setup involves two asterisk servers - both running 1.4.9.

Basically what happens is that only one channel is being recorded (hence you’ll only here one person talking - the callee). Here are the details:

Call comes into PRI/Tenor which sends to Asterisk #1, call is sent to Asterisk #2 with dialplan, if call is answered using SIP phone on Asterisk #2 then both sides of call record properly. If call is answered via external phone which is dialed by Asterisk #2 sending request to Asterisk #1 and then out PRI only the external phone side is monitored and not the in side.

If using MixMonitor the call appears to record all the way through with dead air for the IN side, but if using Monitor to record then the IN stops as soon as the call is answered. The OUT side records all the way through. All you hear on the IN side is up until the call is answered, so it does record for a little while, but the channel is ending early for some reason.

Thanks in advance.

A little more notes on this. The system was working fine and suddenly failed but wasn’t noticed until some time later so we don’t know what changed to cause the problem. The change happened around Nov 28th and wasn’t noticed until about 2 weeks later. Also, there is a 3rd Asterisk which does exactly the same job as the 2nd Asterisk for recording and it was working perfectly the whole time. This leads me to believe the problem is on the Asterisk #2 and not Asterisk #1.

With that said and trying to not contradict it to badly, we just removed the nat=no on Asterisk #1 in the sip.conf referring to the Asterisk #2 server and now part of the problem is resolved. Here’s what is listed in the Asterisk #1 sip.conf:
host={URL is listed here}
;nat=no ; This was causing one way audio as the IN was ending once the call was picked up by a cell or other land line

What has been resolved is the calls now record both sides of the conversation if answered by either a SIP phone or land line or cell. What is still not resolved is a little more complicated and not mentioned earlier, so I will explain in more detail…

We can call Asterisk #3 for free with our cell, then pass the call to Asterisk #2 and dial out Asterisk #1 to another land line… thus avoiding to many charges on our cells and having the ability to record our calls. However, as of around Nov 28 when the problems started, these calls now only record the person answering and not the person making the call. If we call Asterisk #3 and then out #1 it records fine. If we call Asterisk #2 and then out #1 it also records fine. Unfortunately this is not what we want due to A) it cost us to call #2 directly, and B) we want to record on #2 and not #3.

So, any ideas where to start looking as to why we can’t call Asterisk #3, pass it to #2 and then out #1 to the land lines and record both sides of the call? Again, it all worked before Nov 28 but we don’t know what changed then. All these servers are on high speed connections with direct IP addresses and not behind any routers/NAT/firewalls other than what might be running on their own systems.

Thanks in advance.