Hi

 

Hoping someone can point me in the right direction.

I have a Kamailio Ver: 4.2.3-1.1  running in front of a few asterisk servers Ver: 13.17.2  sip is routed to an asterisk server depending the domain name in the sip request, all working as expected . but if a call is put on hold  after resuming the call the party that placed the call on hold can’t hear any audio. The other party can hear . do I need to do anything special to handle re-invites for calls put on hold?

 

 

Gerry Kernan

 

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