<DIV>Hi all.</DIV>
<DIV> </DIV>
<DIV>I have a problem to hear one side, when the second is working fine.</DIV>
<DIV> </DIV>
<DIV>softphone -> ser -> asterisk (IVR) -> extension in IVR -> ser -> pstn -> regular phone.</DIV>
<DIV> </DIV>
<DIV>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.</DIV>
<DIV> </DIV>
<DIV>here is the log what i am receiving:</DIV>
<DIV> </DIV>
<DIV><BR>9 headers, 9 lines<BR>Found RTP audio format 8<BR>Found RTP audio format 101<BR>Peer audio RTP is at port xxx.xxx.xxx.xxx:27232<BR>Found description format telephone-event<BR>Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)<BR>Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)<BR>set_destination: Parsing <sip:phonenumber@realm;ftag=as4783926c;lr=on> for address/port to send to<BR>set_destination: set destination to serserverip, port 5060</DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV>inside sip.conf </DIV>
<DIV> </DIV>
<DIV>disallow=all <BR>allow=ulaw <BR>allow=alaw</DIV>
<DIV> </DIV>
<DIV>now my soft phone using G729,G723,alaw</DIV>
<DIV> </DIV>
<DIV>Any help will be more than appreciated. </DIV><p>
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