Callers reporting first part of greeting cut off

Hello, we received several reports from our customers that the first part of our telephone greeting was cut off when they called in. For example, if we typically say “Sally and Joe’s Hair Salon” when we receive telephone calls, the caller would probably hear something like “Joe’s Hair Salon.”

The system is an 8-port fxo module from xorcom for the incoming lines. The endpoints are aastra 9143i (but I have also tested with the 53i).

Does anyone have any ideas for how to fix this problem (other than the obvious answer of picking up the phone and pausing momentarily before greeting the caller).

Thanks,
Jason

Hi, I still have not been able to figure this problem out… but I enabled call recording on the freepbx/asterisk server to try and eliminate the phone as the source of the problem.

The call recording sounds perfect. However, I think I remember reading somewhere that there are two modes for call recording. One that is good for monitoring agents and one that is good for monitoring the quality of a trunk. I’m guessing that the mode that is good for monitoring agents would not discard late packets whereas the one that monitors trunks would discard late packets so that what is in the recording more closely matches what was heard on the actual phone. Does anyone know if this is true and if so, how it can be configured?

I have received some advice that the phone may be doing a codec negotiation when you first pick it up that is causing the delay. I did a sip debug, but I didn’t notice any delay, then again, I’m not sure what I’m looking for exactly.

The theory that I came up with on my own is that it may have to do with the aastra xml scripts that the phones are using. The phones appear to have several visits to various aastra php files when various things happen (like a call is answered).

It is somewhat difficult for me to test the problem thoroughly with one person and the small after-hours window I have, but I believe I have also been able to notice that if the extension that answers the call is part of a ring-group the problem is evident, but If I were to have the inbound route go directly to one phone, the problem would not be evident.

As a test of the aastra xml theory, I have already eliminated all of the php files that the phone I’m testing with accesses upon various events. After doing that, I didn’t notice any improvement to the problem.

I would really like to know if my perfect call recording means that there is no delay with the phone to asterisk communication/initial negotiation or if there is some other mode I can put call recording in to make the recordings sound just as they did on the phones.

Another theory that I tested is that it might have to do with echo cancellation or echo training. I lowered the echotraining and disabled the echo cancellation and neither provided any improvement.

Thanks,
Jason

I will assume you’re talking about an incoming call from outside not hearing the first part of the greeting. I find that I have to start recording my greeting, wait for about 2 seconds then say the greeting. It takes Asterisk a second or two to establish the talk path, especially when SIP endpoints are involved. I’m a little surprised that there is a delay with the Xorcom but don’t know what sort of hardware or even what Asterisk version you are running on.

I would like to see a dialplan mod of “answer” and then “wait(1)” before the voicemail greeting begins. This would give Asterisk time to cut through all talk paths. In the absence of that, just have your users record a 1 or 2 second dead space before they start speaking their greeting.

Hi, this problem might be with the voicemail greeting to, but the specific problem I’m dealing with is when a person answers an incoming call. The first part of what they say is cut off.