i have one problem. If i calling CiscoIP->Ast1->Ast2->Provider, and provider send to Ast2 that Client not unavailable in this time, Ast1 easy continue fake ringing to Cisco.
If i call from CiscoIP direct to ->Ast2->Prov, that all its good, in the phone i hear that, client unavailable in this time, please call back letter.
Please help my, why ast1 not accept right answer from ast2.
ast1 - sip.conf
;CiscoIP in ast 1 and last the same, only context another name.
[100]
type=friend
host=dynamic
username=100
secret=1234
nat=yes
canreinvite=no
context=trans
directmedia=yes
disallow=all
allow=alaw
== Using SIP RTP CoS mark 5
– Executing [420XXXXXX@AST2:1] Dial(“SIP/AST1-0000bf8b”, “OOH323/420XXXXXX@ ISP1”) in new stack
-- Called OOH323/420XXXXXX@ ISP1
== Everyone is busy/congested at this time (1:0/1/0)
– Executing [420XXXXXX@AST2:2] Dial(“SIP/AST1-0000bf8b”, “SIP/420XXXXXX@ ISP2”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/420XXXXXX@ ISP2
– SIP/tc150119-0000bf8d is ringing
– SIP/tc150119-0000bf8d is making progress passing it to SIP/AST1-0000bf8b
yes, but my questions its, that asterisk 1 send to the phone normal ring tone, if then you can see, in the log going error code. If i call direct touch asterisk 2, that waiting for answer and, if will be error, than going to hangup phone. Im not understanding, why asterisk 1 not waiting from asterisk 2 answer, and send immedlity fake ring.
I am having difficult understanding your English, there are a lot of clearly wrong words, but I can’t work out what all of them should be.
Also, you haven’t provided enough logging to be really clear what is happening. You need “sip set debug on” on Asterisk 2.
However, the log you have provided shows two calls. The first call is via ISP1. The second call is via ISP2.
ISP1 has responded with an error final status, with on intermediate status, except for 100 Trying.
The log shows no final status from ISP2, but they have sent:
180 Ringing
followed by:
183 Progress
Typically in such a case, the 183 Progress is followed by early media, either consisting of ringback tone, or an error indication, which may be a tone, or a voice announcement. These tones will be forwarded to the caller, but not interpreted by Asterisk. The call will stay in an early media state until either ISP2 sends a final status, or the caller cancels the call.