[asterisk-users] No audio one way

2008-01-30 Thread David Guarnido

Hi,

I have one problem, i´ve a trunk sip Asterisk--- Cisco 2600. Call
inbound work very good, but call outbound don´t work. Call progress but no
audio. Canreinvite=no , no Nat, No problem Codec.

Any idea???

Thanks in advance,
D




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[Asterisk-Users] Trouble Chan Spy

2006-02-24 Thread David Guarnido








Hi list,

 

I got a question:

 

When I try to ChanSpy a SIP channel I only listen one
channel, for example,

 

I call from 302 extension and I have two active channels:

 

SIP/r1-voip-1b7b   
(None)  
Up  Bridged Call(SIP/302-f1f1)

SIP/302-f1f1
[EMAIL PROTECTED] Up  Dial(SIP/[EMAIL PROTECTED]|4

 

When I try to spy this call from another extension:

 

1.SIP/301-fecc
[EMAIL PROTECTED] Up  ChanSpy(SIP/302)

2.SIP/r1-voip-1b7b   
(None)  
Up  Bridged Call(SIP/302-f1f1)

3.SIP/302-f1f1
[EMAIL PROTECTED] Up  Dial(SIP/[EMAIL PROTECTED]|4

 

I got 3 active channels, the one spying, the one that
places the call and the one that receives the call.

My problem is in the spying channel I can only hear
the one that receives the call (3) but I cannot hear the channel (2):

 

Thanks for your help,






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[Asterisk-Users] Problem ChanSpy

2006-03-07 Thread David Guarnido








 

 

Hi list,

 

I got a question:

 

When I try to ChanSpy a
SIP channel I only listen one channel, for example,

 

I call from 302
extension and I have two active channels:

 

SIP/r1-voip-1b7b   
(None)  
Up  Bridged Call(SIP/302-f1f1)

SIP/302-f1f1
[EMAIL PROTECTED] Up 
Dial(SIP/[EMAIL PROTECTED]|4

 

When I try to spy this
call from another extension:

 

1.SIP/301-fecc
[EMAIL PROTECTED] Up  ChanSpy(SIP/302)

2.SIP/r1-voip-1b7b   
(None)  
Up  Bridged Call(SIP/302-f1f1)

3.SIP/302-f1f1
[EMAIL PROTECTED] Up 
Dial(SIP/[EMAIL PROTECTED]|4

 

I got 3 active channels,
the one spying, the one that places the call and the one that receives the
call.

My problem is in the
spying channel I can only hear the one that receives the call (3) but I cannot
hear the channel (2):

 

 In the file sip.conf
: 

 

[302]

canreinvite=yes

[301]

canreinvite=yes

 

 

 

Thanks for your help,






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[Asterisk-Users] Problem ChanSpy

2006-03-08 Thread David Guarnido








Sorry, This is a mistake, sip.conf:

 

[302]canreinvite=no [301]canreinvite=no  Any idea? Thanks     

 

 

 






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