I have a SIP phone at home behind a NAT router registered with an * box at my office with a routable static IP address running version SVN-branch-1.6.0-r175638M.
If I make a call from my SIP phone out a PRI circuit to my cell phone everything works as expected. I hear audio in both directions and all is good. If from the same SIP phone I make a call via our Veracity SIP account to my cell phone I hear no audio in either direction. In trying to find out what is wrong I used tcpdump to see if I could learn anything. I can see the phone sending fixed length UDP packets on to my home network heading to the IP address of the * box. If I run tcpdump on the * box I do not see the packets being received. I do not see the * box sending any packets to my home network either. I have not checked if the * box is receiving packets from Veracity I only know that no audio packets are sent to my home network. If I use tcpdump to watch the SIP phone call via the PRI circuit I see packets both on my home network and my * box. If I use a SIP phone located in my office and make a call via Veracity everything is okay. Also a co-worker has a vpn router on his home network connected to the office vpn server and he can make calls from his SIP phone via Veracity without problems. I can also call his SIP phone from my SIP phone and packets pass as expected. It seems as if audio packets from my SIP phone disappear only if they are involved with a call via Veracity. Does anyone have some idea what I might look at to find what is causing this problem? -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
