Hello!
Just warning you I do not speak English, in General, the essence of the problem.
The essence of the problem. If forwarding is not RTP traffic, if you use the function Answer Dial before RTP traffic goes.
Example config:
exten => r,1,Dial(${TRN}/+7xxxxxxxxxx,300,ttT)
same => n,Hangup()
So there is no RTP traffic, respectively, and the silence. If you add before Dial function Answer all the rules.
Example of console output with RTP debug enabled
[root@voip asterisk]# asterisk-rvvvvvv
Asterisk 14.2.1, Copyright © 1999 - 2016, Digium, Inc. and others.
Created by Mark Spencer markster@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for detail
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.
14.2.1 Connected to Asterisk currently running on voip (pid = 26264)
== Using SIP CoS mark 5 TRN
– Executing [r@tula:1] Dial(“SIP/RTK-0000002a”, “SIP/TRN/+7xxxxxxxxxx,300,ttT”) in new stack
== Using SIP CoS mark 5 TRN
– Called SIP/TRN/+7xxxxxxxxxx
– SIP/RTK-0000002b is making progress passing it to SIP/TRN-0000002a
– SIP/RTK-0000002b is ringing
– SIP/RTK-0000002b is ringing
– SIP/RTK-0000002b answered SIP/TRN-0000002a
– Channel SIP/RTK-0000002b joined ‘simple_bridge’ basic-bridge
– Channel SIP/RTK-0000002a joined ‘simple_bridge’ basic-bridge
– Channel SIP/RTK-0000002b left ‘simple_bridge’ basic-bridge
– Channel SIP/RTK-0000002a left ‘simple_bridge’ basic-bridge
== Spawn extension (tula, r, 1) exited non-zero on ‘SIP/TRN-0000002a’
Number +7xxxxxxxxxx changed. As the name trunk.
Actually the question, who may be faced with this, what could be the reason?