VoipBuster don't Hangup a call!


#1

Hi, My name is Luis Oliveira,

I’m from Portugal and I’m software Developer at AGORA.

I try to make interoperability between Asterisk and VoipBuster and the outgoing calls works right.

But, exist one problem. When I hangup the call in my mobile phone I don’t receive the packet BYE in my asterisk for this reason in my asterisk the call don’t terminate.

trunk configuration
[voipbuster]
type = peer
host=sip1.voipbuster.com
host = 77.72.169.134
username = ****
fromuser = *****
secret = *****
qualify=yes
nat=yes

I think the problem is because I use the function Originate for establish the call between two users. Do you have some sugestion for solve this issue?

Thanks for your attention.

Best Regards Luis Oliveira.


#2

Originate command is not the issue I have done the same task before without any issues, If a BYE request is sent the recipient of the request must confirm the BYE with a 200 (OK) response, which terminates the session and the BYE transaction.

make a sip trace please.


#3

Either you are sending a bad Contact header (incorrect NAT traversal setup) or the ITSP is broken.


#4

Just as an aside, in current version of Asterisk nat=yes is deprecated, use nat=force_rport,comedia instead.


#5

Is deprecated because you are supposed to think about the individual options. For example, part of the problem with one other thread is likely to be that everyone is using comedia, so no-one is prepared to send any media to allow the other side to determine the media address.

If the current case is NAT issue, neither of these options are likely to be the problem. The problem is likely to be the lack of a means of determining the external address, such that Asterisk is sending a private address in the Contact header.