Headcache with one-way audio case.
- Server A Elastix 2.4.0-1 (Asterisk 184.108.40.206) / TDM410P wth FXS module.
. IAX2 trunk to
- Server B: idem as server A
All extensions are SIP.
Using OpenG729/alaw/ulaw for codecs
Server A FXS module act as TELCO interface for a Alcatel 4200E PABX (Fr).
. When a sip@serverA call sip@serverB
. When a sip@serverB call analog phone on FXS/PABX/AnalogPhone
Fail (one-way audio):
. When analog phone take a line through FXS/ServerA and call sip@ServerB
- Call process pass
- Analog can ear sip callee but sip@serverB can not ear analog caller
Test purpose: Puttin’ a analog phone directly to the FXS port (by-pass PABX): All OK
Failure seems to be a mismatch parameters between FXS and PABX carrier signaling.
I’ve ear that in some case, FXS should be tuned to match with France settings (50Hz instead of default 20Hz carrier frequency) when acting with France preset PABX?