Ok. Then Asterisk must see the external IP and think that the device is not behind NAT. A way of "tricking" Asterisk is to set up the customers external IP as localnet on Asterisk and see if that fixes it. If this is the case time for a new router....
-----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Schneur Rosenberg Sent: Monday, January 30, 2012 16:15 To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] No audio on some routers with PAP2T yes net is set to yes on both On Mon, Jan 30, 2012 at 4:08 PM, Dovid Bender <[email protected]> wrote: > Shnuer, > > The issue is that when the packet comes from behind the Baudtech Asterisk > see's an external IP and there for thinks there is no NAT and is looking for > the RTP on the ports stated. When you send from the Linksys it see's a local > IP and knows there is NAT. Do you have nat=yes in sip.conf in both the > general section and for the IP of OpenSIpS? > > Rergards, > > dOvid > > > -----Original Message----- > From: [email protected] > [mailto:[email protected]] On Behalf Of Schneur Rosenberg > Sent: Monday, January 30, 2012 15:07 > To: OpenSIPS users mailling list > Subject: [OpenSIPS-Users] No audio on some routers with PAP2T > > Hi, I have a openSIPS server setup to do registration and load > balancing between 2 Asterisk servers, the Asterisk servers do > everything besides registration and they are load balanced by the > openSIPS servers, incoming calls hit the openSIPS server which sends > it to the Asterisk server and if it needs to go to a local phone it > sends it back to openSIPS where the phone is registered to, outgoing > calls get sent to Asterisk via load balancing and asterisk completes > the call. > > I have a problem with some ata's (in my case pap2t) that when its > behind certain routers (in my case a Baudtech) there is no audio, when > I try a different router it does work, also when I try a different ata > like a spa2102 it does work, also when I connect the pap2t directly to > the asterisk it works fine, NONE of the routers have SIP ALG enabled, > it seems that the nat blocks the audio when the media is from a > different server. > > The interesting thing is that the Baudtech router changes the internal > IP's to the external ip, the other router does not, does that mean > that there is some kind of ALG built into the Baudtech router? even if > it does how come the Asterisk server handles the audio fine while the > openSIPS breaks the audio > > here is a trace of the initial INVITE from the Baudtech (the > problematic one) as you can see the ip at the Via and Contact and in > the c tag in the RTP have been replaced by the router > > U 46.116.60.131:5060 -> 64.69.33.43:5060 > INVITE sip:[email protected] SIP/2.0. > Via: SIP/2.0/UDP 46.116.60.131:5060;branch=z9hG4bK-278a1ace;rport. > From: <sip:[email protected]>;tag=aea4a93b350d6746o0. > To: <sip:[email protected]>. > Call-ID: [email protected]. > CSeq: 101 INVITE. > Max-Forwards: 70. > Contact: <sip:[email protected]:5060>. > Expires: 240. > User-Agent: Linksys/PAP2T-5.1.6(LS). > Content-Length: 442. > Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. > Supported: x-sipura, replaces. > Content-Type: application/sdp. > . > v=0. > o=- 50374 50374 IN IP4 46.116.60.131. > s=-. > c=IN IP4 46.116.60.131. > t=0 0. > m=audio 16476 RTP/AVP 0 2 4 8 18 96 97 98 100 101. > a=rtpmap:0 PCMU/8000. > a=rtpmap:2 G726-32/8000. > a=rtpmap:4 G723/8000. > a=rtpmap:8 PCMA/8000. > a=rtpmap:18 G729a/8000. > a=rtpmap:96 G726-40/8000. > a=rtpmap:97 G726-24/8000. > a=rtpmap:98 G726-16/8000. > a=rtpmap:100 NSE/8000. > a=fmtp:100 192-193. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-15. > a=ptime:30. > a=sendrecv. > > Here is the same invite when send from the DLINK router (here audio is fine) > > U 85.250.89.78:5060 -> 64.69.33.43:5060 > INVITE sip:[email protected] SIP/2.0. > Via: SIP/2.0/UDP 192.168.2.100:5060;branch=z9hG4bK-65cff5ef;rport. > From: <sip:[email protected]>;tag=2ab40bee91703297o0. > To: <sip:[email protected]>. > Call-ID: [email protected]. > CSeq: 101 INVITE. > Max-Forwards: 70. > Contact: <sip:[email protected]:5060>. > Expires: 240. > User-Agent: Linksys/PAP2T-5.1.6(LS). > Content-Length: 442. > Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. > Supported: x-sipura, replaces. > Content-Type: application/sdp. > . > v=0. > o=- 19246 19246 IN IP4 192.168.2.100. > s=-. > c=IN IP4 192.168.2.100. > t=0 0. > m=audio 16438 RTP/AVP 0 2 4 8 18 96 97 98 100 101. > a=rtpmap:0 PCMU/8000. > a=rtpmap:2 G726-32/8000. > a=rtpmap:4 G723/8000. > a=rtpmap:8 PCMA/8000. > a=rtpmap:18 G729a/8000. > a=rtpmap:96 G726-40/8000. > a=rtpmap:97 G726-24/8000. > a=rtpmap:98 G726-16/8000. > a=rtpmap:100 NSE/8000. > a=fmtp:100 192-193. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-15. > a=ptime:30. > a=sendrecv. > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
