Hello everyone! I have Asterisk server and cisco ip phone 7965 and all file of sip protocol and cnf.xml and xml was created with information, In this time i can make call from zoiper or 3cx software to cisco phone without any problem and i can listen between them, But the problem when I need call from cisco phone to zoiper or 3cx I cant because direct the line go to like disconnect the line by beep!!! What the problem or how can I solve it please !??
Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
> 0x7f50600235d0 – Strict RTP learning after remote address set to: zoiper ip
Are you using the sip or pjsip channel technologies?
You need to provide the protocol logging for whichever of the above you are using.
You may also need to provide the relevant configuration file, if you want more than general hints at what is wrong.
Note Asterisk doesn’t send beeps as the result of a failed call, unless there is explicit dialplan to do so, or the called party sends one in the media stream. It will simply send a SIP failure response, and the phone may or may not generate some form of sound as a result.
As you say you are using chan_sip, we need the output from “sip set debug on” and the contents of sip.conf and anything that it includes, in as much as it is in the general section or relates to any of the parties in the call.
In this time I solved some problem, Now i can make call from cisco to zoiper or other one like 3cx, So If i call from cisco to zoiper I can listen inside of zoiper perfect but cisco phone I cant listen !!! why ? I dont now!!
So in the same idea but if I call from cisco phone to 3cx the line between them its very good without any problem because in to sides can listen and Speak up between them, I tried to check the codec in server and zoiper and I enable all of zoiper codec inside of server but the problem not solve??
Any help about the new problem?
I am tried now so the result:
In cisco phone when I dial to zoiper the cisco open the line but the connection is still open in cisco and the other party is not contacted or ring after 1 mint. the line go to disconnected !!!
Also note that nat= does not enable NAT handling in Asterisk; what it does is to enable workarounds for some cases where NAT is not properly handled, in particular, where it is not properly handled by the remote end. It doesn’t make Asterisk handle NAT properly when Asterisk is inside NAT.
The IP addresses in the logs are either obfuscated or being misused, so the logs don’t tell us whether NAT is actually being used.
The same as before, but on a call that is failing in the new way. Also you need to explain the 1.1.1. in your addresses. It is particularly important to distinguish between public and private addresses.
For the network structure, there is no command; you need to understand how your network is put together.