I have created two accounts in sip.conf file and registered with the dialer
Let the first account is A and the second one is B
1,When I dial extension for B, I am getting error 488 not acceptable here
2,When B dials the extension for A, I get the incoming call and when I answer the call it gets disconnected automatically
For (2) kindly see the logs below
== Using SIP RTP CoS mark 5
– Executing [8001@users:1] Dial(“SIP/abdul-00000004”, “SIP/arun,50”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/arun
– SIP/arun-00000005 is ringing
[Apr 30 20:48:26] ERROR[27387][C-00000042]: netsock2.c:305 ast_sockaddr_resolve: getaddrinfo(“0122ec0cI4”, “(null)”, …): Temporary failure in name resolution
[Apr 30 20:48:26] WARNING[27387][C-00000042]: chan_sip.c:10914 process_sdp_c: Unable to lookup RTP Audio host in c= line, ‘IN IP4 0122ec0cI4’
[Apr 30 20:48:26] WARNING[27387][C-00000042]: chan_sip.c:10475 process_sdp: Insufficient information in SDP (c=)…
== Everyone is busy/congested at this time (1:0/0/1)
– Auto fallthrough, channel ‘SIP/abdul-00000004’ status is ‘CHANUNAVAIL’
Please help me to solve this as I am very new to asterisk