TIPS Probation Passed – setting RTP source address to xx.x.x.x

[safa0786]

Hello everyone my setup is:-


Asterisk = ONLINE | Dahdi = ONLINE | MySQL = ONLINE x
SSH = ONLINE | Apache = ONLINE | Iptables = ONLINE x
Fail2ban = ONLINE | Internet = ONLINE | Ip6Tables = ONLINE x
Disk Free = ADEQUATE| Mem Free = ADEQUATE| NTPD = ONLINE x
SendMail = ONLINE | Samba = ONLINE | Webmin = ONLINE x
Ethernet0 = ONLINE | Ethernet1 = N/A | Wlan0 = N/A x
PIAF Installed Version = 2.0.6.5 under *HARDWARE* x
FreePBX Version = 2.11.0.35 x
Running Asterisk Version = 11.8.1 x
Asterisk Source Version = 11.8.1 x
Dahdi Source Version = 2.9.0 x
Libpri Source Version = 1.4.14 x
IP Address = 192.168.X.X on eth0 x
Operating System = CentOS release 6.5 (Final) x
Kernel Version = 2.6.32-431.1.2.0.1.el6.x86_64 - 64 Bit x
Incredible Version = 11.8

On connection to an incoming call via PSTN where asterisk 11.8.1 is Dialing an internal extension on answering the call there is about 6-7 seconds before audio is heard on either side. When looking at the CLI traces when I answer the incoming call that asterisk extensions were dialing, I see immediately upon answering — Probation passed – setting RTP source address to x.x.x.x( Endpoint IP Answered the Call). It appears to be an issue is that the RTP link(audio) setup is delayed.
i have tried adding strictrtp=yes in rtp.conf but no luck

Anyone have suggestions on how to fix this issue

 

[atsak]

This sounds like an early media issue, but the carriers I use don't have it so I am not sure what to do to fix it. Try googling no audio for a few seconds asterisk (or freepbx or pbxinaflash) and what solutions have been offered