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22.7832
2009 Jun 10
1
Resetting Marker Bits
Hi All,
I'm looking for how to enable SIP Markers, or specifically, how to have
the TIME reset when a call route changes.
I'm debugging an issue, where a sip client we have switching to
one-way-audio, when an asterisk server fruther down the call path dials
out to the PSTN. Scenario is:
SIP Client -> A*k1 -> A*k2 -> PSTN Provider/Gradwell -> O2 ->
Mobile