Asterisk - RTP - SIP -Disconnected

Zaraab

Well-known member
Zaraab - Well-known member  
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<<<-------
 

GenXOutsourcing

Active member
GenXOutsourcing - Active member  
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?
 

Zaraab

Well-known member
Zaraab - Well-known member  
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

Active member
GenXOutsourcing - Active member  
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
 

Zaraab

Well-known member
Zaraab - Well-known member  
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?
 
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