I have a problem with not being able to receive incoming phone calls. To give some background: incoming phone calls work if I set “Allow Anonymous Inbound SIP Calls” to yes. I have an inbound route set with my DID. In addition, I’ve tried clearing all the INCOMING settings, to no avail. And as with some other posts, I have the same issue that Asterisk can’t seem to identify the peer. I’ve been tearing my hair out on this and have searched these forums and couldn’t seem to find an answer that fit my problem. I used the trunk settings provided to me by the SIP provider (Switch2Voip.us). Help would be greatly appreciated! Thank you in advance! Below are my trunk settings and the CLI output:
OUTGOING Settings:
PEER Details:
username=[USERNAME]
type=peer
secret=[PASSWORD]
progressinband=never
port=5060
nat=auto
insecure=very
ignoresdpversion=yes
host=sip.switch2voip.us
dtmfmode=rfc2833
context=from-trunk
canreinvite=no
allow=g729&g711&g723
fromuser=+13053963482
INCOMING Settings:
USER Context: [USERNAME] [I’ve also tried the DID number]
USER Details:
username=[USERNAME]
password=[PASSWORD]
type=user
port=5060
dtmfmode=rfc2833
pogressinband=never
context=from-pstn
allow=g729&g711&g723
And below is the output on the CLI:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [13053963482@from-sip-external:1] NoOp(“SIP/66.33.146.52-0000001d”, “Received incoming SIP connection from unknown peer to 13053963482”) in new stack
– Executing [13053963482@from-sip-external:2] Set(“SIP/66.33.146.52-0000001d”, “DID=13053963482”) in new stack
– Executing [13053963482@from-sip-external:3] Goto(“SIP/66.33.146.52-0000001d”, “s,1”) in new stack
– Goto (from-sip-external,s,1)
– Executing [s@from-sip-external:1] GotoIf(“SIP/66.33.146.52-0000001d”, “0?checklang:noanonymous”) in new stack
– Goto (from-sip-external,s,5)
– Executing [s@from-sip-external:5] Set(“SIP/66.33.146.52-0000001d”, “TIMEOUT(absolute)=15”) in new stack
– Channel will hangup at 2014-05-02 16:40:46.381 PDT.
– Executing [s@from-sip-external:6] Log(“SIP/66.33.146.52-0000001d”, "WARNING,“Rejecting unknown SIP connection from 66.33.146.52"”) in new stack
– Executing [s@from-sip-external:7] Answer(“SIP/66.33.146.52-0000001d”, “”) in new stack
0x7f10f00060b0 – Probation passed - setting RTP source address to 66.33.166.133:20064
– Executing [s@from-sip-external:8] Wait(“SIP/66.33.146.52-0000001d”, “2”) in new stack
– Executing [s@from-sip-external:9] Playback(“SIP/66.33.146.52-0000001d”, “ss-noservice”) in new stack
– <SIP/66.33.146.52-0000001d> Playing ‘ss-noservice.ulaw’ (language ‘en’)
– Executing [s@from-sip-external:10] PlayTones(“SIP/66.33.146.52-0000001d”, “congestion”) in new stack
– Executing [s@from-sip-external:11] Congestion(“SIP/66.33.146.52-0000001d”, “5”) in new stack
== Spawn extension (from-sip-external, s, 11) exited non-zero on ‘SIP/66.33.146.52-0000001d’
– Executing [h@from-sip-external:1] Hangup(“SIP/66.33.146.52-0000001d”, “”) in new stack
== Spawn extension (from-sip-external, h, 1) exited non-zero on ‘SIP/66.33.146.52-0000001d’