Route DID from freePBX to CUCM

Hi,
I want to route incoming DID from Asterisk to CUCM. On Asterisk I have 100Did’s, two of them need to be routed to CUCM. SIP trunk btw asterisk and cucm is UP. On asterisk inound route for DID 5101586 I have set destination (SIP trunk to CUCM).
Asterisk trunk details:
host=172.31.252.205
type=friend
canreinvite=no
insecure=port,invite
qualify=yes
nat=no
disallow=all
allow=ulaw&alaw
trustrpid=yes
sendrpid=pai
context=from-trunk

When I call 5101586 from my mobile070303024 I didnt get to CUCM.

Log from the call:

Connected to Asterisk 1.8.24.0-8 currently running on MiSwitch (pid = 2261)
Verbosity is at least 6
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [[email protected]:1] Set(“SIP/VipONE-000043fb”, “GROUP()=OUT_3”) in new stack
– Executing [[email protected]:2] Goto(“SIP/VipONE-000043fb”, “from-trunk,5101586,1”) in new stack
– Goto (from-trunk,5101586,1)
– Executing [[email protected]:1] Set(“SIP/VipONE-000043fb”, “__FROM_DID=5101586”) in new stack
– Executing [[email protected]:2] Gosub(“SIP/VipONE-000043fb”, “app-blacklist-check,s,1()”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/VipONE-000043fb”, “0?blacklisted”) in new stack
– Executing [[email protected]:2] Set(“SIP/VipONE-000043fb”, “CALLED_BLACKLIST=1”) in new stack
– Executing [[email protected]:3] Return(“SIP/VipONE-000043fb”, “”) in new stack
– Executing [[email protected]:3] ExecIf(“SIP/VipONE-000043fb”, “0 ?Set(CALLERID(name)=070303024)”) in new stack
– Executing [[email protected]:4] Set(“SIP/VipONE-000043fb”, “__CALLINGPRES_SV=allowed_not_screened”) in new stack
– Executing [[email protected]:5] Set(“SIP/VipONE-000043fb”, “CALLERPRES()=allowed_not_screened”) in new stack
– Executing [[email protected]:6] Goto(“SIP/VipONE-000043fb”, “ext-trunk,7,1”) in new stack
– Goto (ext-trunk,7,1)
– Executing [[email protected]:1] Set(“SIP/VipONE-000043fb”, “TDIAL_STRING=SIP/KonSI”) in new stack
– Executing [[email protected]:2] Set(“SIP/VipONE-000043fb”, “DIAL_TRUNK=7”) in new stack
– Executing [[email protected]:3] Goto(“SIP/VipONE-000043fb”, “ext-trunk,tdial,1”) in new stack
– Goto (ext-trunk,tdial,1)
– Executing [[email protected]:1] Set(“SIP/VipONE-000043fb”, “OUTBOUND_GROUP=OUT_7”) in new stack
– Executing [[email protected]:2] GotoIf(“SIP/VipONE-000043fb”, “1?nomax”) in new stack
– Goto (ext-trunk,tdial,4)
– Executing [[email protected]:4] ExecIf(“SIP/VipONE-000043fb”, “1?Set(CALLERPRES()=allowed_not_screened)”) in new stack
– Executing [[email protected]:5] Set(“SIP/VipONE-000043fb”, “DIAL_NUMBER=5101586”) in new stack
– Executing [[email protected]:6] GosubIf(“SIP/VipONE-000043fb”, “0?sub-flp-7,s,1”) in new stack
– Executing [[email protected]:7] Set(“SIP/VipONE-000043fb”, “OUTNUM=5101586”) in new stack
– Executing [[email protected]:8] Dial(“SIP/VipONE-000043fb”, “SIP/KonSI/5101586,300,”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Called SIP/KonSI/5101586
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [[email protected]:9] Set(“SIP/VipONE-000043fb”, “CALLERID(number)=070303024”) in new stack
– Executing [[email protected]:10] Set(“SIP/VipONE-000043fb”, “CALLERID(name)=070303024”) in new stack
– Executing [[email protected]:11] Hangup(“SIP/VipONE-000043fb”, “”) in new stack
== Spawn extension (ext-trunk, tdial, 11) exited non-zero on ‘SIP/VipONE-000043fb’

Can you please advice how to configure above?

Thanks.
Regards.

Well not sure but should look at the stage of

– Executing [[email protected]:8] Dial(“SIP/VipONE-000043fb”, “SIP/KonSI/5101586,300,”) in new stack

Assuming.
DDI —tdial ----------FreePBX ---- KonSIP --------CUCM
As no details of network schematic if CUCM are on the same network it assume problem becomes from KonSI and CUCM so may be by capturing packets could get more explicit details why call couldn’t progress.

Hi Ricardo,

Thanks for your reply.
FreePBX and CUCM are connected throw VPN.
Here you can find capture from the call
https://drive.google.com/file/d/1Zfv9ExaaVp6VGN4EKx7w4ILsuW5Pb2k9/view

Thanks.
Regards

Hi, thanks.

I have downloaded file but it refer to Vlan network traffic. Capture, not SIP.

By the way have you try to change type= to peer?

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