Hi,
I am using Asterisk with various brands and models of SIP phones. Especially the Welltech phones LP201 are particularly nasty with volume and echo. Even with the input gain (microphone) of the Welltech set to the max, the PSTN end can hardly hear the SIP user on incoming calls. Ztmonitor also only gives a level of around 3 === from the SIP phone.
I have to increase the rxgain and txgain by about 4 - 7, but then all my other phones are so loud that it distorts and the echo-canceller can't compensate (on outgoing calls only). The ringing noise also is at full level (deafening loud).
I have also noticed that incoming calls from PSTN into the TE405P to SIP are amplified differently than outgoing calls from SIP to PSTN. It seems the TXgain is not used on incoming calls...
Are my observations above expected, or is there something wrong with the code? Is auto-gain implementable / recommendable so that all the SIP phones will sound the same (volume-wise) to the outside PSTN user, and vice versa ?
Could we build this into the sip configuration so that individual gain per phone is adjustable if needed ?
Here is a cut-out of my zapata.conf (just in case I am really stupid :-)
[channels] context=default switchtype=euroisdn signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=no transfer=yes cancallforward=yes echocancel=yes echotraining=yes echocancelwhenbridged=yes rxgain=5.0 txgain=7.0 immediate=no ; Channels inherit configuration above them ; Span 1 group=1 context=default signalling=pri_cpe channel => 1-15 channel => 17-31
; Span 2 group=2 context=ppms signalling=pri_cpe channel => 32-46 channel => 48-62
; Span 3 group=3 context=default signalling=pri_cpe channel => 63-77 channel => 79-93
; Span 4 group=4 context=default signalling=pri_net channel => 94-108 channel => 110-124
Hope anyone can shed some light on this. I have been breaking my head on this about 4 days now, trying just about anything...
Thanks
Walter Klomp
Singapore.
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