Does anyone have experience using ShoreTel SIP trunks to integrate an Asterisk system?
I am having trouble when the ShoreTel system transfers an incoming call from a SIP trunk to the voicemail system. From the SIP traffic, it looks like it negotiates a codec correctly, but once the RTP stream starts the call drops or there is no audio. I see errors in Asterisk such as: chan_sip.c:1944 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 104 (Critical Request) Has anyone run into this before or have any ideas? Thanks, Joe _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
