[asterisk-users] Advices on how to evaluate voice quality in a mixed Dahdi/SIP environment ?
oza.4h07 at gmail.com
Wed May 18 09:44:25 CDT 2016
I've got the following setup:
PSTN ---- ITSP ---- SDSL Modem-Router --<SIP ?>-- Gateway --<BRI>---
Asterisk with B410P --- SIP Phones
Both SDSL Modem-Router and Gateway are managed by my ITSP.
Some calls coming from PSTN and forwarded to an other PSTN number have a
poor voice quality.
How can I best illustrate this ?
A friend advised me to simply record incoming DAHDI channel, for instance.
How can I then translate record WAV file into meaningful figures ?
More generaly, what would you suggest ?
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