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
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