Increase TX and RX in individual SIP extensions?

General Help
1

I am looking to increase the TX for one SIP extension used for a softphone. Is this possible? When I increase the overall TX, my softphone is flawless but the analog phones wont quite work right so i just want to adjust this one extension.

I am using a Digium 24 Port analog card for my trunks and analog extensions. I have no digital trunks or VOIP.

I am not going to upgrade my version of FreePBX until this one bites the dust so please do not suggest that. The rest of the stats are below in my sig.

Thanks!

2

You have no signature.

3

D’oh. I did on the old format.

FreePBX 2.10.1.6

4

There’s no way I’m the only one wanting this.

5

I believe that the Dahdi “Helper module” is not that “per channel” granular as yet , you can however do that in the old fashioned way, maybe disable the dahdi module and start with:-

http://www.voip-info.org/wiki/view/DAHDI

6

Not DAHDi… SIP…

7

Exactly. I see there is a volume command but I wouldn’t know where to put it (see below).

-= Info about function ‘VOLUME’ =-

[Synopsis]
Set the TX or RX volume of a channel.

[Description]
The VOLUME function can be used to increase or decrease the ‘tx’ or ‘rx’ gain
of any channel.
For example:
Set(VOLUME(TX)=3)
Set(VOLUME(RX)=2)
Set(VOLUME(TX,p)=3)
Set(VOLUME(RX,p)=3>

[Syntax]
VOLUME(direction[,options])

[Arguments]
direction
Must be ‘TX’ or ‘RX’.
options
p: Enable DTMF volume control

8

He has only analog trunks and SIP phones should never need TX/RX “normalizing” it doesn’t work that way, ergo, he needs to balance his trunk’s RX/TX (and perhaps fxotune them while he is at it) individually for best effect.

9

If they don’t need normalizing then why is there a command provided by Asterisk to do just that.

If you read the post he only wants to do it for one single phone. Not for everyone.

10

Agreed again. i did the tuning and everything works well except one SIP extension, hence this post.

11

If everything other than one SIP extension is fine then surely it must be the SIP extension that is behaving badly, try another soft phone maybe, the asterisk volume function is channel based and would be complicated to implement in FreePBX for just that one extension.

12

I have tried several soft phone application and devices.

It would make sense to be able to adjust each extension to satisfy all users, don’t you think?

13

Perhaps so, but I have personally never found that necessary, SIP and IAX2 normally “just work”, FXO’s on the other hand often need balancing individually above and beyond fxotune which although done on individual channels, just handle the impedence matching of them into the PSTN. dahdi_monitor performed on the various channels against either a voice call or preferably a milliwatt() application on both ends will either confirm or disprove that they need adjusting., either globally or individually. Just Try it :-

channel=1;echo -e “Channel: DAHDI/$channel/13139379996\nApplication: milliwatt” > /var/spool/asterisk/outgoing/call.file;dahdi_monitor $channel -vv

If each channel is identically balanced (about 75% on the visual linear scale) and all the RX/TX gains on all the channels are matched within ±100 or so then I apologize for wasting your time.