vicidial.org

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<<<-------

Why are you using dahdi 2.7? Try 3.1 and also change your stun server in rtp.conf away from Google

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<<<-------

This happens when asterisk does not receive a response from the phone....

Do you have the rtp ports open in firewall?
Did this just start? or is a new install?

You didnt say how you install? what OS?

carpenox wrote:Why are you using dahdi 2.7? Try 3.1 and also change your stun server in rtp.conf away from Google

The dahdi version 3 has compilation error in centOS

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?

ah yes for cent 7 u need 2.11

Yeah... Any idea why this error is coming? #nox

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

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

For getting the ring, the SIP phone must send signal(id/pass) and verify itself with asterisk and then its able to get the call is not it? So where & how the response is missing as for RTP towards the vici?

Open ports 10000 - 20000 for udp

carpenox wrote:Open ports 10000 - 20000 for udp

How to do this? using iptables?

firewall-cmd i believe is in cent 7, i know for sure its in cent 8

carpenox wrote:Open ports 10000 - 20000 for udp

all are open. It was issue with the vpn i was using.

dhijrwn wrote:

carpenox wrote:Open ports 10000 - 20000 for udp

How to do this? using iptables?

Avoid using Iptables and use the framework---> Firewalld

isnt that what i said zaraab? firewall-cmd is firewalld fyi :-p

carpenox wrote:isnt that what i said zaraab? firewall-cmd is firewalld fyi :-p

absolutely my friend!

Just was guiding him that its firewalld other than iptables!