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Zaraab wrote:Why is this happening? -------> Disconnecting call for lack of RTP activity in 61 seconds
If this were to happen for any port(UDP/TCP) issue, then why at first place the SIP phone will be able to register?
---->>>Asterisk 13.29.2 / Latest SVN/ Dahdi 2.7.0.1<<<-------
carpenox wrote:Why are you using dahdi 2.7? Try 3.1 and also change your stun server in rtp.conf away from Google
Zaraab wrote:Hey #nox. The dahdi 3.1 has compilation error in centos 7
I have scratch installed and firewall is setup with all ports open and running #Genx
The sip phone is ringing.. Which means the phone is sending response to asterisk right?
GenXOutsourcing wrote:Zaraab wrote:Hey #nox. The dahdi 3.1 has compilation error in centos 7
I have scratch installed and firewall is setup with all ports open and running #Genx
The sip phone is ringing.. Which means the phone is sending response to asterisk right?
Ringing means that Vici can send to the phone............. but the phone needs to respond rtp to stay connected
carpenox wrote:Open ports 10000 - 20000 for udp
carpenox wrote:Open ports 10000 - 20000 for udp
dhijrwn wrote:carpenox wrote:Open ports 10000 - 20000 for udp
How to do this? using iptables?
carpenox wrote:isnt that what i said zaraab? firewall-cmd is firewalld fyi :-p
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