FIX: Aastra Update Broke Speakerphone

Stewart

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I have a tale of 3 different installations. The first hasn't had any updates for a few months and is running fine. The second has most of the updates (Freepbx, scripts, etc.) including the 2.6.0.1008 firmware for the 57i phones and is running fine. The ones that have all the updates including the new Aastra scripts are the ones experiencing a speakerphone problem.

The problem is that it's as if the gain it turned up +30. Or in otherwords, as if the microphone is in their mouth. If they stand 12 feet away, they may sould distant by the reverb, but at least they're clear. I can reproduce it on all 4 systems I've put the new scripts on. If I set:
handsfree tx gain: -15
things are much better. Though the documentation says it only goes to +/-10, I've read online that it goes to +/-15 and there is even a difference if I put in -30, so I'm not sure.

This affects all of my Aastra phones, but none of my Polycom phones. All in all there are about 40 Aastra phones and 30 Polycoms. All of the Aastra's are affected and None of the Polycoms are. Handset and Sidetone are fine. Rx in the speakerphone is fine. It's just the gain of the mic for the speakerphone.

Any ideas?
 
When your making the TX RX gain adjustments are you seeing any effect after a phone reboot? Are you putting the adjustments in the right file with the right syntax? If you role back the firmware does the problem go away? The scripts don't effect the volume but the firmware and configs can.
 
Yes. When I make the change in the aastra.cfg file and reboot the phone the change is noticeable. The adjustments are going in the proper place since it really is making a sizeable difference. The problem is that I need to make any adjustments at all. I would think that going +/- 3-6 would be all I need to tweak it for the users satisfaction, but that leaving the default config alone should be enough for most users. That's the way it has been up until this point. I have other systems using the same 2.6.0.1008 firmware without issue. The only difference on the systems with the problem is the new scripts. I wouldn't expect them to have any bearing at all, but does the installation modify any other file that would do this? I see that a new update to the firmware has been released. I could try it on Monday and see what happens. The mac.cfg files are the same as they were before the update and are auto-generated by the log-in script.

Also, when I'm on the Aastra handset and there is no conversation I hear a slight hiss or some form of background noise. It's not loud at all and is perfectly fine until I speak into the phone. Then the sound goes away and comes back when I'm done talking. I'm guessing it's some form of CNG, but I don't know. I dont really care if it's there or not, but it's pretty annoying that it comes and goes. I'm not sure where to even begin on that one and haven't really even looked into it.
 
I think I would do three things if I had this problem, the first is I'd call Aastra the second is I'd build a small but identical system that I could test one of these phones with at the customer site the third is I'd roll back the firmware.

There are some debug type things that you can do with these phones but I haven't had a need to use them.

Normally the 57i needs a reduction in gain as they run a little "hot" on the audio but as you've indicated -4 should be about right.

How are calls between extensions?

Are these SIP lines or DAHDI connected pots lines?
 
I thought we had to go through our vendor to get support. I'll call them on Monday. I do have an identical system at our office and we are able to reproduce the issue. It doesn't matter if it's an internal or external call and all calls are SIP.
 
Well, I rolled back to a previous version and installed the new scripts again and no more problems. Hmmm..... I wonder what the problem really was...

I called Aastra yesterday and they said there was nothing they could do. The only adjustment to make was the "handsfree tx gain" setting and if that didn't fix it, that was all I could try. I did notice that if I transfered a call to a Polycom phone and then transfered it back to the Aastra phone it sounded OK. In case someone else has the problem and wants to carry the ball.....
 
Sadly, the problem has resurfaced. And again, the only change was the upgrade to the new scripts. <Sigh> Am I seriously the only one with this problem?
 
There's a reason they call it Bleeding Edge. :deathb:
 
While I sat scratching my head, I realized 2 things:

1-The In-Call button to park a call was gone.
2-The only change was that I was calling from a different phone on the system than from when it was working fine.

So this morning I tried making making calls on both the phone that was working earlier and the phone I had just tried that wasn't working. Bingo! The first one was still working properly and the second was really whacked out! In to the MAC.cfg's we go...

A couple of things that I noticed in the config of the bad phone:

1. The sprecode entry was blank where before it was "asterisk;500". Once I put that back in, the park button re-emerged.

2. There is a new entry called "sip customized codec". Since it wasn't there before, I commented it out. (While I was at it, I commented every new line out but have since put them back in.) After a reboot, the audio sounded normal again. I limited it to the portion that read "payload=9;ptime=20;silsupp=on". Once removed, it's working normal. (Funny, here it says it should be ptime=30. Can someone help out with an explanation?) Since payload=9 is for G.722 and Asterisk 1.4 has trouble with G.722 I yanked it and things are good now.

Some informed commenting would be helpful.
 

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