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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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. >BenBen - Yes. http://lists.digium.com/pipermail/asterisk-users/2004-March/039663.html JT