No audio on extension calls


#1

I am new to Asterisk and decided to install Asterisk @ Home. I setup 2 SIP extensions and got X-lite on 2 of my PC’s. I got them logged in and attempted to make some calls. I get ringing and can answer the calls, but there is no audio. I have checked to make sure that I do not have windows firewall running on either machine.

I do not have port forwarding setup on my router, but I do not think that should be an issue since both PCs and the asterisk server are all on the same switch on the internal side of the firewall.

Eferything from the PCs looks good. They are sending out voice packets to the Asterisk server, but nothing is comming from the server. I pulled the debug log and this is where it looks like the audio is not being sent:

May 18 22:40:22 VERBOSE[3258] logger.c: – SIP/2000-75d0 answered SIP/2001-6b7f
May 18 22:40:22 VERBOSE[3258] logger.c: – Attempting native bridge of SIP/2001-6b7f and SIP/2000-75d0
May 18 22:40:22 DEBUG[2449] chan_sip.c: Stopping retransmission on ‘584C9F34-26C2-4687-9FAF-9453DA81E80E@192.168.1.100’ of Response 28885: Match Found
May 18 22:40:38 DEBUG[3258] channel.c: Bridge stops because we’re zombie or need a soft hangup: c0=SIP/2001-6b7f, c1=SIP/2000-75d0, flags: No,No,No,Yes
May 18 22:40:38 DEBUG[3258] channel.c: Bridge stops bridging channels SIP/2001-6b7f and SIP/2000-75d0
May 18 22:40:38 DEBUG[3258] chan_sip.c: update_call_counter(2000) - decrement call limit counter
May 18 22:40:38 DEBUG[3258] app_dial.c: Exiting with DIALSTATUS=ANSWER.

Any help would be greatly appreciated.


#2

Seems to be a dialplan issue. Post your extensions.conf


#3

I was searching for a solution and decided to verify if I had the most recent verion of A@H. It seems that there has been some new releases since I originally downloaded it. I decided to do a fresh install of A@H 2.8. I added my test phones and everything came up working. Now on to playing with the other features.

Thank you very much for your help.