I am having problem diagnosing a call problem. On both a Cisco phone and a Linksys 942 I am only getting one side of the call when connected over a WAN link or internet connection. I have set nat=yes and qualify in sip.conf and the phone registers fine. I can hear the other end, but they do not hear anything, no voice or dtmf. I found a tip about changing the RTP rate from .03 to .02 on Sipura phones to match Asterisk rate and did that. I also made sure the RTP range for the phone and the server was set to 10000 thru 20000. These phones work fine when on the same subnet as the server. The server shows the following message: NOTICE[24975]: rtp.c:330 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 192.168.10.56 But I have silence suppresion off on the linksys phone. -- Bruce Nortex Networks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060510/ffce989c/attachment.htm
I have contiuned to fight this problem all day, and still have not found a solution. I did get DTMF tones to the other end, but no voice. Any tips on where to look? On 5/10/06, Bruce Reeves <asterisk@nortex-networks.com> wrote:> > I am having problem diagnosing a call problem. On both a Cisco phone and a > Linksys 942 I am only getting one side of the call when connected over a WAN > link or internet connection. I have set nat=yes and qualify in sip.confand the phone registers fine. I can hear the other end, but they do not hear > anything, no voice or dtmf. I found a tip about changing the RTP rate from > .03 to .02 on Sipura phones to match Asterisk rate and did that. I also made > sure the RTP range for the phone and the server was set to 10000 thru 20000. > These phones work fine when on the same subnet as the server. The server > shows the following message: > > NOTICE[24975]: rtp.c:330 process_rfc3389: Comfort noise support incomplete > in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: > 192.168.10.56 > > But I have silence suppresion off on the linksys phone. > > -- > Bruce > Nortex Networks >-- Bruce Nortex Networks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060510/7a80fccf/attachment.htm
Hi! I found, that there is 4 options for nat: -no -never -yes -always no and never is ok but sometimes yes, and sometimes always worked for me :-o> I am having problem diagnosing a call problem. On both a Cisco phone and a > Linksys 942 I am only getting one side of the call when connected over a WAN > link or internet connection. I have set nat=yes and qualify in sip.conf and > the phone registers fine. I can hear the other end, but they do not hear > anything, no voice or dtmf. I found a tip about changing the RTP rate from > .03 to .02 on Sipura phones to match Asterisk rate and did that. I also made > sure the RTP range for the phone and the server was set to 10000 thru 20000. > These phones work fine when on the same subnet as the server. The server > shows the following message:-- WoodOO-[P]an[G]alaktikan[A]gent-People <][> http://shadow.pganet.com wpeople@shadow.pganet.com]iCQ#33118021[wpeople.on.iRCNet]wpeople@RedHat.users