I have a number if Linksys SPA942 Sip phones connected to my Asterisk PBX, and all agents use the same Plantronics M110 headsets. Even with the headset volume at maximum I often hear agents telling callers “I can barely hear you”. I have played with the phone’s “Headset Input Gain:” and “Headset Additional Input Gain:” settings but nothing seems to help much. Perhaps those setting refer to the headset mic and not the headset speaker. I’ve tried one or two other headset models but those were no better. Does anyone have any advice?
Just out of curiousity,
How do the calls sounds without the headsets? Are the headsets plugged into the Handset phone portion, or is it connecting to the small RCA output port on the side of the SPA942?
Let me know. Thanks.
Handset calls are fine - the headsets are connected to the 2.5mm headset jack.
I am starting to believe that there it has to deal more with the Plantronics Headsets. I know this sounds dumb, but is your volume on your phone all the way up by using the (+) sign on the keypad? i can help you increase the gain of calls that are inbound, but it will make handset calls louder and probably annoying. I have used that 2.5mm jack beofre with a 2-way speaker system, here to find out that we needed an amplifier to boost the sounds since this port doesn’t have one but the + and - volume control. I would suggest using some type of softphone solution for your computer such as Xlite and use the computers volume to adjust with the headsets, and just using the SPA942’s for other calls. Then again, How do the calls sound on the headsets from extension to extension? is it just calls from outside that sound like this? Are you running full SIP or do you have any PSTN lines? Let me know. thanks.
Hi again. Yes the phones’ volume controls are turned up all the way. All of our calls come in a DiDs from a SIP provider and run through the Asterisk box. Internal calls seems ok. The low volume only seems to happen on calls that come in on the SIP DiDs… not on all of the calls, but on many of them.
Depending on which version of asterisk you have, v.1.4 and 1.6 i’ve tinkered with. There is a file called chan_dahdi.conf in the /etc/asterisk/ folder. Open this file up using vi editor or one of your liking, and you will find the following:
If you cannot find it, search through the file. In Vi, to search for words or text in the document you would type /rxgain and hit enter, to go to the next word in the search hit “n” on the keyboard. Once you find this, modify the txgain up 6 db by doing txgain=6.0. rxgain is the volume that your phone sends to asterisk, txgain is the volume that is sent to your phone from asterisk. once your changes have been made, if you used the vi editor, type the following to save and quit:
hit esc on the keyboard, type :wq and hit enter. :wq writes and quits. Then you will need to restart asterisk everytime a change is made if you need to go back in to adjust it more. See if this helps, and let me know. Thanks.
Did this issue get resolved?
I’m still running 1.2. I tried 1.4 awhile back but had many issues. I have a couple questions about your proposed solution:
Am I correct that instead of chan_dahdi.conf I should edit zapata.conf?
Do chan_dahdi.conf or zapata.conf have any effect on calls to SIP phones? My Asterisk box has no zap cards, but zaptel and ztdummy modules are loaded. Please let me know and thanks for your interest.
You will be modifying Zapata.conf, and yes, modifying this file will also affect SIP connections. In here you will also find other modifications such as detectfax and others. I would suggest trying the solution below. Just make a copy of the zapata.conf file if needed, and make your changes. If it doesn’t solve your problem, then I would undo the changes and we can go from there. Good Luck!!!
P.s. Haha, I’m intrested in all the posts because we ran into so many problems with our customer’s installation that it made me want to help others that never receive replys back from anyone. I guess you can say, your loss if my gain, but if we both figure it out, it’s 2 more people that know the resolution and can start helping others.
My calls are not coming in on a PSTN line… they come in over SIP from my DiD provider. I’m having trouble understanding how zapata.conf is involved in these purely SIP calls. I’ll give your solution a try, but can you help me understand why zapata.conf affects SIP calls?
To answe your question, I’m not sure. All i know is that asterisk does funny things. I’m still wondering why, when I use the gui, does it make changes to my extensions.conf file and add characters in places that shouldn’t have them. My only other suggestion if this doesn’t work, is to consult your SIP providor for the DID’s and ask them if they can up their gain so you can hear end users more clearly. Do you know what codec your using?