Hi Has anyone experienced any issues with calls through asterisk server with a netted phone connected to a Cisco 18XX series router.
I'm experiencing one way audio when the caller calls from the phone connected to the asterisk server to the outside world (via a SIP provider). It's the audio in to the caller that is failing. We're using asterisk 1.8.7.0 When looking at the logs I'm seeing RTP transmission errors as the RTP is being router to port 0 on the correct IP address. If I do a SIP trace I can see that all SIP traffice is being router to the correct port on the IP address. I just can't work out where this port 0 is coming from and have a suspicion that the router is doing something funny. Has anyone experienced anything like this before? Thanks in Advance Ish -- Ishfaq Malik <i...@pack-net.co.uk> Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _____________________________________________________________________ -- 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
