[asterisk-users] ChanSpy problems in Unicall to SIP calls

[Guillermo Freige]

Hi:
I'm having audio dropouts in ChanSpy when the call is originated in a 
Unicall (E1/MFCR2) channel and the destination is an Agent using a SIP 
phone. If the agent is using a traditional phone (going from the PBX to 
asterisk via another Unicall line) no dropouts are present. The dropouts 
seems to occur only in the audio coming from the Unicall line, and mostly 
when there is no audio coming from the SIP end. If both are talking, or only 
the SIP end, audio seems fine.
I saw the problem in all asterisk 1.2 versions, but after the late ChanSpy  
rewriting, it seemed the problem was solved in an unloaded test system, but 
in the production one, the dropouts are still present.
Any idea?

Guillermo Freige

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