[SR-Users] Audio stops after resuming call from hold

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 Infinity IT | 17 The Mall | Beacon Court | Sandyford | Dublin D18 E3C8 | Ireland Tel: +353 - (0)1 - 293 0090 | E-Mail: <mailto:gerry.kernan@infinityit.ie> gerry.kernan(a)infinityit.ie Managed IT Services Infinity IT - <http://www.infinityit.ie/> www.infinityit.ie IP Telephony Asterisk Consulting - <http://www.asteriskconsulting.com> www.asteriskconsulting.com Contact Centre Total Interact - <http://www.totalinteract.com> www.totalinteract.com