[Asterisk-Users] Some audio problems

[Alex]

Hi all.
 
I have a problem to hear one side, when the second is working fine.
 
softphone  ->  ser -> asterisk (IVR) -> extension in IVR -> ser -> pstn -> regular phone.
 
The audio which coming from regular phone i can hear without problem, but the audio from softphone i can not hear in the regular phone.
 
here is the log what i am receiving:
 

9 headers, 9 lines
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port xxx.xxx.xxx.xxx:27232
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)
set_destination: Parsing <sip:phonenumber at realm;ftag=as4783926c;lr=on> for address/port to send to
set_destination: set destination to serserverip, port 5060
 
 
inside sip.conf 
 
disallow=all                   
allow=ulaw                    
allow=alaw
 
now my soft phone using G729,G723,alaw
 
Any help will be more than appreciated. 

		
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