[Asterisk-Users] Asterisk in pass-thru mode

[John Todd]

At 8:13 PM +0800 on 4/15/04, Radius wrote:
>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

Ben -
   Yes.

http://lists.digium.com/pipermail/asterisk-users/2004-March/039663.html

JT