Displaying 1 result from an estimated 1 matches for "088f93e0".
2009 Apr 27
1
Packet2packet bridging while in sip.conf canreinvite=no
...sk to be in the middle of the RTP-stream so he can
provide MusiconHold and so...
Now, what the Asterisk CLI tells me when I make a call from my one
internal SIP-phone to another internal SIP-phone is :
Verbosity is at least 25
== Spawn extension (intern, 51, 1) exited non-zero on
'SIP/BT201-088f93e0'
-- Executing [52 at intern:1] Dial("SIP/GXP1200-088f93e0", "SIP/BT201|
30") in new stack
-- Called BT201
-- SIP/BT201-088faa00 is ringing
-- SIP/BT201-088faa00 answered SIP/GXP1200-088f93e0
-- Packet2Packet bridging SIP/GXP1200-088f93e0 and
SIP/BT201-088...