[asterisk-users] Dropped call on new CISCO router for no reason!
Hello Everyone, Just getting in a new cisco router, and would really like to get it up and running as soon as possible. Everything is configured from what we can see. This is a NAT setup. After 2 seconds on a successfully established call we reach retrans max, and asterisk disconnects the call. We have no idea why this is happening. SIP and RTP is flowing as expected. Your help is greatly appreciated, Nick. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropped call on new CISCO router for no reason!
This is a classic symptom of having reinvites and/or direct media enabled on Asterisk or SIP ALG enabled on the router. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Cameo Sent: Monday, January 06, 2014 9:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dropped call on new CISCO router for no reason! Hello Everyone, Just getting in a new cisco router, and would really like to get it up and running as soon as possible. Everything is configured from what we can see. This is a NAT setup. After 2 seconds on a successfully established call we reach retrans max, and asterisk disconnects the call. We have no idea why this is happening. SIP and RTP is flowing as expected. Your help is greatly appreciated, Nick. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropped call on new CISCO router for no reason!
Hello Eric, I knew this problem all so well however, never knew CISCO sip alg was enabled by default. The following settings got us up and going shortly after the email: no ip nat service sip udp port 5060 ip nat inside source static udp 192.168.2.5 5060 interface Dialer0 5060 access-list 130 permit udp any any range 8000 65535 route-map voip-rtp permit 1 match ip address 130 ip nat inside source static PRIVATE IP PUBLIC IP route-map voip-rtp Happy New Year to All, Nick from Toronto. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropped call on new CISCO router for no reason!
On 14-01-06 09:27 AM, Nick Cameo wrote: Hello Everyone, Just getting in a new cisco router, and would really like to get it up and running as soon as possible. Everything is configured from what we can see. This is a NAT setup. After 2 seconds on a successfully established call we reach retrans max, and asterisk disconnects the call. We have no idea why this is happening. SIP and RTP is flowing as expected. Your help is greatly appreciated, Nick. Show us the problem, give us a SIP trace[1]. [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
