Connected to Asterisk 1.0.9 currently running on ast (pid = 2660)
Verbosity is at least 55
-- Executing Dial("SIP/713-3f9a", "SIP/14024645971@isphone|30") in new stack
-- Called 14024645971@isphone
-- SIP/isphone-a54e is making progress passing it to SIP/713-3f9a
-- SIP/isphone-a54e is ringing
Nov 9 18:44:38 NOTICE[2661]: rtp.c:298 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible
-- SIP/isphone-a54e answered SIP/713-3f9a
-- Attempting native bridge of SIP/713-3f9a and SIP/isphone-a54e
Nov 9 18:44:49 NOTICE[2661]: chan_sip.c:2792 process_sdp: No compatible codecs!
== Spawn extension (default, 14024645971, 1) exited non-zero on 'SIP/713-3f9a'
ast*CLI>