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 (ContextExtensionPri ) State Appl. Data
SIP/5001-c60b (company11 ) Up Bridged
Call SIP/1234-faf1
SIP/1234-faf1 (company1 5001 1 ) Up Dial
SIP/5001|20|r
2 active channel(s)
sip*CLI> sip show channels
Peer User/ANRCall ID Seq (Tx/Rx) Lag Jitter Format
192.168.1.1015001257684717aa 00104/0 0ms ms ULAW
210.17.211.5 1234003094c2-fd 00104/00102 0ms ms ULAW
2 active SIP channel(s)
Thanks.
Ben
Ben -
Yes.
http://lists.digium.com/pipermail/asterisk-users/2004-March/039663.html
JT
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