[asterisk-dev] Working with SIP Communicator or disabling reINVITE

[Samir S]

Hello All,

Does anyone have experience of interworking Asterisk with the SIP
communicator ?
It seems that the Sip communicator does not support reINVITE
Is it possible to disable this generate of reINVITE in Asterisk and make it
default to a pure proxy (for testing purposes only) ?

Any help would be greatly appreciated.

Thanks in advance,
Samir
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