Hi every one
I have two server asterisk, I use server A as a Sip server, Server B as a proxy server
I register server 2 extension:1000 and 2000 on server A, when I call from 2000 to 1000, I want call follow like this: 2000–> server A—> server B → server A—> 1000
but when I call server B have notice: Failed to authenticate on INVITE to '“2000” sip:2000@172.17.17.52;tag=as7007d314’
** – SIP/172.17.17.54-000001ac is circuit-busy**
** Everyone is busy/congested at this time (1:0/1/0)**
** – Auto fallthrough, channel ‘SIP/2000-000001ab’ status is ‘CONGESTION’**
This is my config on 2 server
type, for the phones, should be peer, not friend. This will cause a problem if the caller ID matches on the outgoing call matches a device name on the remote system. (I believe PJSIP doesn’t have this distinction, so it is important to avoid a caller ID/device name clash.)
insecure=invite does nothing without secret=. insecure=port is rarely needed. canreinvite is a deprecated form of directmedia. It is very rare that you need to force nat to no.
Hi David
45 í time wait i set up for a call
I change my paradigm to test
From server B call to server A ok
From server A call to server B note ok
when i use sngrep tool to check i see sip invite from server A havan’t line:Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSGE, OPTIONS, INFO, SUBSCRIBE but invite from server B to A have this line