Hello Kindly, I need your assist as I have installed chan_ss7 with asterisk and got no voice on calls although asterisk is working fine With sip I am using -sangoma A104de -dahdi 2.6.6 -Asterisk 1.6.2 -chan_ss7 2.2.0 Also I get *NOTICE[22933]: mtp.c:1993 mtp_thread_main: Full dahdi input buffer detected, incoming packets may have been lost on link 'l1' (count=65.)* At the beginning of an incoming call is it related to my case? You can find my config files attached -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-ss7/attachments/20130618/b723c4da/attachment.htm> -------------- next part -------------- A non-text attachment was scrubbed... Name: ss7.conf Type: application/octet-stream Size: 4440 bytes Desc: not available URL: <http://lists.digium.com/pipermail/asterisk-ss7/attachments/20130618/b723c4da/attachment.obj> -------------- next part -------------- A non-text attachment was scrubbed... Name: system.conf Type: application/octet-stream Size: 818 bytes Desc: not available URL: <http://lists.digium.com/pipermail/asterisk-ss7/attachments/20130618/b723c4da/attachment-0001.obj>