I think it’s starting to work! Well my asterisk is trying to forward the call to cucm but cucm don’t accept it… lol So I guess it a problem with cucm, I’ll try some settings.
Let’s say my cellphone number is is 450-555-5555
and My DID number from voip.ms is 450-888-8888
When I try to call my DID from my cellphone I get this :
-- Executing [4508888888@from-trunk:1] NoOp("SIP/voipms-00000026", "Catch-All DID Match - Found 4508888888 - You probably want a DID for this.") in new stack
-- Executing [4508888888@from-trunk:2] Goto("SIP/voipms-00000026", "ext-did,s,1") in new stack
-- Goto (ext-did,s,1)
-- Executing [s@ext-did:1] Set("SIP/voipms-00000026", "__FROM_DID=s") in new stack
-- Executing [s@ext-did:2] Gosub("SIP/voipms-00000026", "app-blacklist-check,s,1") in new stack
-- Executing [s@app-blacklist-check:1] GotoIf("SIP/voipms-00000026", "0?blacklisted") in new stack
-- Executing [s@app-blacklist-check:2] Set("SIP/voipms-00000026", "CALLED_BLACKLIST=1") in new stack
-- Executing [s@app-blacklist-check:3] Return("SIP/voipms-00000026", "") in new stack
-- Executing [s@ext-did:3] ExecIf("SIP/voipms-00000026", "0 ?Set(CALLERID(name)=4505555555)") in new stack
-- Executing [s@ext-did:4] Set("SIP/voipms-00000026", "__CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing [s@ext-did:5] Set("SIP/voipms-00000026", "CALLERPRES()=allowed_not_screened") in new stack
-- Executing [s@ext-did:6] Goto("SIP/voipms-00000026", "ext-trunk,2,1") in new stack
-- Goto (ext-trunk,2,1)
-- Executing [2@ext-trunk:1] Set("SIP/voipms-00000026", "TDIAL_STRING=SIP/cucm01") in new stack
-- Executing [2@ext-trunk:2] Set("SIP/voipms-00000026", "DIAL_TRUNK=2") in new stack
-- Executing [2@ext-trunk:3] Goto("SIP/voipms-00000026", "ext-trunk,tdial,1") in new stack
-- Goto (ext-trunk,tdial,1)
-- Executing [tdial@ext-trunk:1] Set("SIP/voipms-00000026", "OUTBOUND_GROUP=OUT_2") in new stack
-- Executing [tdial@ext-trunk:2] GotoIf("SIP/voipms-00000026", "1?nomax") in new stack
-- Goto (ext-trunk,tdial,4)
-- Executing [tdial@ext-trunk:4] ExecIf("SIP/voipms-00000026", "1?Set(CALLERPRES()=allowed_not_screened)") in new stack
-- Executing [tdial@ext-trunk:5] Set("SIP/voipms-00000026", "DIAL_NUMBER=s") in new stack
-- Executing [tdial@ext-trunk:6] GosubIf("SIP/voipms-00000026", "0?sub-flp-2,s,1") in new stack
-- Executing [tdial@ext-trunk:7] Set("SIP/voipms-00000026", "OUTNUM=s") in new stack
-- Executing [tdial@ext-trunk:8] Dial("SIP/voipms-00000026", "SIP/cucm01/s,300,") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called cucm01/s
-- SIP/cucm01-00000027 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [tdial@ext-trunk:9] Set("SIP/voipms-00000026", "CALLERID(number)=4505555555") in new stack
-- Executing [tdial@ext-trunk:10] Set("SIP/voipms-00000026", "CALLERID(name)=Rogers Wilcot") in new stack
-- Executing [tdial@ext-trunk:11] Hangup("SIP/voipms-00000026", "") in new stack
== Spawn extension (ext-trunk, tdial, 11) exited non-zero on 'SIP/voipms-00000026'
== Manager 'admin' logged on from 127.0.0.1
== Manager 'admin' logged off from 127.0.0.1
== Manager 'admin' logged on from 127.0.0.1
== Manager 'admin' logged off from 127.0.0.1
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [4508888888@from-trunk:1] NoOp("SIP/voipms-00000028", "Catch-All DID Match - Found 4508888888 - You probably want a DID for this.") in new stack
-- Executing [4508888888@from-trunk:2] Goto("SIP/voipms-00000028", "ext-did,s,1") in new stack
-- Goto (ext-did,s,1)
-- Executing [s@ext-did:1] Set("SIP/voipms-00000028", "__FROM_DID=s") in new stack
-- Executing [s@ext-did:2] Gosub("SIP/voipms-00000028", "app-blacklist-check,s,1") in new stack
-- Executing [s@app-blacklist-check:1] GotoIf("SIP/voipms-00000028", "0?blacklisted") in new stack
-- Executing [s@app-blacklist-check:2] Set("SIP/voipms-00000028", "CALLED_BLACKLIST=1") in new stack
-- Executing [s@app-blacklist-check:3] Return("SIP/voipms-00000028", "") in new stack
-- Executing [s@ext-did:3] ExecIf("SIP/voipms-00000028", "0 ?Set(CALLERID(name)=4505555555)") in new stack
-- Executing [s@ext-did:4] Set("SIP/voipms-00000028", "__CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing [s@ext-did:5] Set("SIP/voipms-00000028", "CALLERPRES()=allowed_not_screened") in new stack
-- Executing [s@ext-did:6] Goto("SIP/voipms-00000028", "ext-trunk,2,1") in new stack
-- Goto (ext-trunk,2,1)
-- Executing [2@ext-trunk:1] Set("SIP/voipms-00000028", "TDIAL_STRING=SIP/cucm01") in new stack
-- Executing [2@ext-trunk:2] Set("SIP/voipms-00000028", "DIAL_TRUNK=2") in new stack
-- Executing [2@ext-trunk:3] Goto("SIP/voipms-00000028", "ext-trunk,tdial,1") in new stack
-- Goto (ext-trunk,tdial,1)
-- Executing [tdial@ext-trunk:1] Set("SIP/voipms-00000028", "OUTBOUND_GROUP=OUT_2") in new stack
-- Executing [tdial@ext-trunk:2] GotoIf("SIP/voipms-00000028", "1?nomax") in new stack
-- Goto (ext-trunk,tdial,4)
-- Executing [tdial@ext-trunk:4] ExecIf("SIP/voipms-00000028", "1?Set(CALLERPRES()=allowed_not_screened)") in new stack
-- Executing [tdial@ext-trunk:5] Set("SIP/voipms-00000028", "DIAL_NUMBER=s") in new stack
-- Executing [tdial@ext-trunk:6] GosubIf("SIP/voipms-00000028", "0?sub-flp-2,s,1") in new stack
-- Executing [tdial@ext-trunk:7] Set("SIP/voipms-00000028", "OUTNUM=s") in new stack
-- Executing [tdial@ext-trunk:8] Dial("SIP/voipms-00000028", "SIP/cucm01/s,300,") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called cucm01/s
-- SIP/cucm01-00000029 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [tdial@ext-trunk:9] Set("SIP/voipms-00000028", "CALLERID(number)=4505555555") in new stack
-- Executing [tdial@ext-trunk:10] Set("SIP/voipms-00000028", "CALLERID(name)=Rogers Wilcot") in new stack
-- Executing [tdial@ext-trunk:11] Hangup("SIP/voipms-00000028", "") in new stack
== Spawn extension (ext-trunk, tdial, 11) exited non-zero on 'SIP/voipms-00000028'
But now when I make a call to my provider from cucm, it won’t work. It says the number I dialed is invalid (message from My Asterisk), so I guess my outbound route is incorrect? But why a call from one of my Asterisk sip extensions (X-lite) work?
EDIT1 : Ah I just found out that Actually all my calls (even from cucm) are forwarded to my cucm (haha) So it just do a loop when I try to place a call… Will check my routes.
Thanks a lot for your time!