[asterisk-users] Asterisk and Kamailio NAT problem
Hello I'm using Asterisk as a SIP client of Kamailio with RTPproxy. Asterisk is behind NAT. X-Lite and SNOM phones behind NAT work fine. But when I try to connect with an Asterisk behind NAT, the Asterisk client doesn't receive sound. I already tried in 2 different NATs, with no firewalls. This is my Asterisk config: [kamailio] type=peer host=xxx.xxx.xxx.xxx disallow=all allow=ulaw allow=alaw allow=gsm allow=g726 qualify=1000 username=my_username fromuser=my_username secret=password sip*CLI sip show peers Name/username HostDyn Nat ACL Port Status kamailio/my_username xxx.xxx.xxx.xxx 5060 OK (890 ms) Is there something missing in my SIP.CONF to improve the compatibility with Kamailio? How can I debug the RTP stream in Asterisk? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- 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
[asterisk-users] Asterisk and Kamailio NAT problem
Hello Im using Asterisk as a SIP client of Kamailio with RTPproxy. Asterisk is behind NAT. X-Lite and SNOM phones behind NAT work fine. But when I try to connect with an Asterisk behind NAT, the Asterisk client doesn't receive sound. I already tried in 2 different NATs, with no firewalls. This is my Asterisk config: [kamailio] type=peer host=xxx.xxx.xxx.xxx disallow=all allow=ulaw allow=alaw allow=gsm allow=g726 qualify=1000 username=my_username fromuser=my_username secret=password sip*CLI sip show peers Name/username HostDyn Nat ACL Port Status kamailio/my_username xxx.xxx.xxx.xxx 5060 OK (890 ms) Is there something missing in my SIP.CONF to improve the compatibility with Kamailio? How can I debug the RTP stream in Asterisk? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- 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
Re: [asterisk-users] Asterisk and Kamailio NAT problem
Try putting nat=yes in your asterisk peer Tom -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joao Gomes Pereira Sent: Monday, July 27, 2009 9:50 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk and Kamailio NAT problem Hello Im using Asterisk as a SIP client of Kamailio with RTPproxy. Asterisk is behind NAT. X-Lite and SNOM phones behind NAT work fine. But when I try to connect with an Asterisk behind NAT, the Asterisk client doesn't receive sound. I already tried in 2 different NATs, with no firewalls. This is my Asterisk config: [kamailio] type=peer host=xxx.xxx.xxx.xxx disallow=all allow=ulaw allow=alaw allow=gsm allow=g726 qualify=1000 username=my_username fromuser=my_username secret=password sip*CLI sip show peers Name/username HostDyn Nat ACL Port Status kamailio/my_username xxx.xxx.xxx.xxx 5060 OK (890 ms) Is there something missing in my SIP.CONF to improve the compatibility with Kamailio? How can I debug the RTP stream in Asterisk? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- 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 ___ -- 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
