>> Aha, I remembered at the PJSIP level a feature[1] got added when the >> Google Voice patch was done. It hasn't been exposed to be explicitly >> configurable in Asterisk though. >> >> [1] >> https://www.pjsip.org/pjsip/docs/html/structpjsip__tpselector.htm#a1622f416d48eb173aed22b750aa28dfa >> > > In fact the flow functionality[1] may get you closer to how you need > things. There was a bug though with it, so either 18 branch needs to be > used or this change[2] included. > > [1] > https://github.com/asterisk/asterisk/blob/master/configs/samples/pjsip.conf.sample#L161 > [2] https://gerrit.asterisk.org/c/asterisk/+/15256
Great! I'm testing it! I'll be back. Thanks Michael -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
