Hello Guys im new to asterisk im trying to do call from asterisk to my provider but im getting this below mention error can anyone help me out to solve this error.
Using SIP RTP CoS mark 5
– Executing [00447714286969@fone:1] NoOp(“SIP/1001-00000024”, “00XXXXXXXXXXXX”) in new stack
– Executing [00447714286969@fone:2] Goto(“SIP/1001-00000024”, “9384650743,00XXXXXXXXXXXX,1”) in new stack
– Goto (9384650743,00XXXXXXXXXXXX,1)
– Executing [00XXXXXXXXXXXX@9384650743:1] Dial(“SIP/1001-00000024”, “SIP/9384650743/00XXXXXXXXXXXX”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/9384650743/00XXXXXXXXXXXX
[Oct 5 14:44:25] WARNING[1342][C-00000018]: chan_sip.c:24054 handle_response_invite: Received response: “Forbidden” from ‘“1001” sip:1001@192.168.2.28;tag=as11e7d521’
== Everyone is busy/congested at this time (1:0/0/1)
– Auto fallthrough, channel ‘SIP/1001-00000024’ status is ‘CHANUNAVAIL’
You are not allowed to use the provider’s service. You will need to use sip set debug, to get an idea as to whether they are rejecting your password or simply rejecting you. They might need fromuser or fromdomain setting.
Thanks that you had replied to solve my error as told me that They might need fromuser or fromdomain yes i had tried & its get worked i had mentioned fromuser insted of this i was using defaultuser which was not working then i had tried fromuser as you informed me & its work im really thanking your for your valuable time & solution.
Also i need one more help from you if you dont mind how to do call recording of extensions weather they dial any number or they receive any call on there extension can you teach me step by step