[Asterisk-Users] Asterisk in pass-thru mode

[Radius]

Hi all,

Below is what I did to run Asterisk in pass-thru mode:

sip.conf:
[general]
disallow=all
allow=ulaw
canreinvite=yes

For each channel, canreinvite=yes is enabled. No dial command has 't' option.

However, it seems that Asterisk still stay in the media path and bridge the 2 end points. Am I missing something??? 


sip*CLI> show channels
        Channel  (Context    Extension    Pri )   State Appl.         Data
SIP/22225001-c60b  (company1                1   )      Up Bridged Call  SIP/1234-faf1
  SIP/1234-faf1  (company1   5001         1   )      Up Dial          SIP/22225001|20|r
2 active channel(s)

sip*CLI> sip show channels
Peer             User/ANR    Call ID      Seq (Tx/Rx)  Lag      Jitter  Format
192.168.1.101    22225001    257684717aa  00104/00000  00000ms  0000ms  ULAW
210.17.211.5     1234        003094c2-fd  00104/00102  00000ms  0000ms  ULAW
2 active SIP channel(s)


Thanks.
Ben
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