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@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. --------------------------------- Do you Yahoo!? Yahoo! Small Business - Try our new resources site! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050323/4e8ecc9a/attachment.htm