I need some help getting outbound calls working via two PBX servers running Asterisk.
The one i have complied from source, the dialplan is successfully calling the SIP peer which is my trunk.
The other PBX is running Elastix.
THE PBX RUNNING FROM SOURCE:
THE PBX RUNNING ELASTIX
Now the peers are up and in an OK status, what makes me believe that it is not the actual trunk config is that i have inbound working from another PBX in our network, i have tried calls out through there also but same result. Both other PBX systems are working 100% - inbound and out. So i think there is something up with the box I have complied from source.
This is the messages i get from asterisk on the failed calls:
“Dial”,“SIP/XYZ”,“2019-07-29 14:47:06”,“2019-07-29 14:47:06”,0,0,“FAILED”,“DOCUMENTATION”,“1564411626.68”,""
Using SIP RTP CoS mark 5
> 0x7fbd5c06c4a0 – Strict RTP learning after remote address set to: IPADDY:41844
– Executing [NUMBER@phones:1] Goto(“SIP/andre-00000052”, “outgoing,0768329443,1”) in new stack
– Goto (outgoing,NUMBER,1)
– Executing [NUMBER@outgoing:1] Dial(“SIP/NAME-00000052”, “TRUNKNAME”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/TRUNKNAME
– Got SIP response 603 “Declined” back from ELASTIXPBX
– SIP/TRUNKNAME is busy
== Everyone is busy/congested at this time (1:1/0/0)
– Auto fallthrough, channel ‘SIP/NAME-00000052’ status is ‘BUSY’