Hello When I make calls from home to the PSTN by going through the Net -> Asterisk -> the Net -> VoIP provider -> PSTN, I get no sound either way. I assume it's because I must tell Asterisk to use fixed ranges of UDP ports and map ports accordingly on the NAT firewall under which it is located on the LAN at work. Here's the schema: home > NAT > Internet > NAT > Asterisk > NAT > Internet > VoIP provide > PSTN > callee I took care of the NAT at home by using fixed ports in X-Lite + used STUN, so I guess the problem is located on the Asterisk side. 1. What are the settings (in sip.conf?) to tell Asterisk to use specific ports for RTP? 2. With this kind of setup, does Asterisk stay in the loop to forward RTP packets, or do X-Lite at home and the VoIP provider send RTP to each other directly? Thank you.
Hey Vincent -> 1. What are the settings (in sip.conf?) to tell Asterisk to use specific > ports for RTP?I guess you didn't see my reply earlier today - that setting is in rtp.conf> 2. With this kind of setup, does Asterisk stay in the loop to forward RTP > packets, or do X-Lite at home and the VoIP provider send RTP to each other > directly?That all depends on your "canreinvite" setting (in sip.conf, per peer or user). If set to no, then asterisk will stay in the picture throughout your call. If set to yes, the call will be passed off so the rtp traffic goes directly from X-lite to the VoIP provider. Keep in mind that if you do have it set to yes, the VoIP provider may not necessarily use the same rtp ports that you want to use. - Noah
Vincent Delporte wrote:> Hello > > When I make calls from home to the PSTN by going through the Net -> > Asterisk -> the Net -> VoIP provider -> PSTN, I get no sound either way. > I assume it's because I must tell Asterisk to use fixed ranges of UDP > ports and map ports accordingly on the NAT firewall under which it is > located on the LAN at work. > > Here's the schema: > home > NAT > Internet > NAT > Asterisk > NAT > Internet > VoIP provide > > PSTN > callee > > I took care of the NAT at home by using fixed ports in X-Lite + used > STUN, so I guess the problem is located on the Asterisk side. > > 1. What are the settings (in sip.conf?) to tell Asterisk to use specific > ports for RTP? > 2. With this kind of setup, does Asterisk stay in the loop to forward > RTP packets, or do X-Lite at home and the VoIP provider send RTP to each > other directly? > > Thank you. > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-userstry rtp.conf :-D -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 189 bytes Desc: OpenPGP digital signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20061130/ee0137b2/signature.pgp