|
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 |
