[Asterisk-Users] Asterisk in pass-thru mode

2004-04-15 Thread Radius 




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. 
DataSIP/5001-c60b  
(company1    
1   )  Up Bridged Call  
SIP/1234-faf1  SIP/1234-faf1  (company1   
5001 1   
)  Up 
Dial  
SIP/5001|20|r2 active channel(s)
sip*CLI> sip show 
channelsPeer 
User/ANR    Call ID  Seq 
(Tx/Rx)  Lag  Jitter  
Format192.168.1.101    5001    
257684717aa  00104/0  0ms  ms  
ULAW210.17.211.5 1234    
003094c2-fd  00104/00102  0ms  ms  ULAW2 active 
SIP channel(s)
 
Thanks.
Ben

Re: [Asterisk-Users] Asterisk in pass-thru mode

2004-04-18 Thread 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  (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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