I have a CVP call flow that is failing with a 503 Service Unaviable message from UCM. I would appreciate any thoughts you might have.
The call arrives at the ingress gateway
The Ingress gateway sends the call to CVP which sends it to ICM
ICM runs a script and generates a label (verified)
The label goes to CVP which hits a static route to the UCM where the agent phone is (no SIP Proxy)
The CVP sends an INVITE to the UCM and I see it arrive on the SIP Trunk by looking at the trace output on UCM
The UCM then sends a 503
The 91919191 ringtone is breifly played before the call is disconnected.
The following is present:
SIP Trunk to UCM set to OnNet and with correct Partition and CSS settings
SIP Trunk to CVP set to OnNet and with correct Partition and CSS settings
Calls can be setup in both directions across the SIP Trunk from UCM to GW
Inbound calls to the SIP Trunk between UCM and CVP seem to be where I get the 503 messages (from UCM)
10.1.1.100 is the UCM
10.1.1.10 is the CVP
10.1.250.1 is the Gateway
Here is the CVP error log
1198: 10.1.1.10: May 30 2010 13:56:36.581 -0700: %CVP_8_0_SIP-3-SIP_CALL_ERROR: CALLGUID = EB0005611000012863D838210A01010A LEGID = EB0005611000012863D838210A01010A-127525299653448 - [OUTBOUND] - DsSipInvitation - <sip:email@example.com;transport=udp>;tag=1885211893 - 1 REJECTED WITH 503 - Service Unavailable Warning: 399 CUCM "Unable to find a device handler for the request received on port 5060 from 10.1.1.10" [id:5004]
1213: 10.1.1.10: May 30 2010 13:56:38.581 -0700: %CVP_8_0_SIP-3-SIP_CALL_ERROR: CALLGUID = EB0005611000012863D838210A01010A LEGID = 37CA21A6-6B6511DF-8065A06E-F17D8643 - [INBOUND] - ABNORMALLY ENDING - SIP code , Reason Hdr [SIP;cause=503] Service Unavailable, GW call using SURV TCL flag [false], NON NORMAL flag [true], USE ERROR REFER flag [true] with AGE (msecs) 31828 and Call History : 811111111127|-1;1001|1; [id:5004]
1220: 10.1.1.10: May 30 2010 13:56:38.675 -0700: %CVP_8_0_SIP-3-SIP_CALL_ERROR: CALLGUID = EB0005611000012863D838210A01010A LEGID = 37CA21A6-6B6511DF-8065A06E-F17D8643 - [INBOUND]: Refer failed with 503 - Service Unavailable. May be a problem with Routing Configuration or Gateway Dial-Peer. [id:5004]
1222: 10.1.1.10: May 30 2010 13:56:38.675 -0700: %CVP_8_0_SIP-3-SIP_CALL_ERROR: CALLGUID = EB0005611000012863D838210A01010A LEGID = 37CA21A6-6B6511DF-8065A06E-F17D8643 - [INBOUND] - ABNORMALLY ENDING - SIP code , Reason Hdr [SIP;cause=503] Service Unavailable, GW call using SURV TCL flag [false], NON NORMAL flag [true], USE ERROR REFER flag [true] with AGE (msecs) 31922 and Call History : 811111111127|-1;1001|1; [id:5004]
I was bringing up a new CVP 8.5 in my lab and had a simple script that did a "Send To VRU", played a bit of music, and released the call. Send to Originator engaged on the NVRU label and the ringback label. Works fine.
Then I changed the script to not release the call but to return the label of a device target of a phone I configured.
Static route in the Outbound Proxy to CUCM. SIP Trunk created to the Proxy in all Default device pools.
I heard a brief ringback and then technical difficulties. The error was the above "Unable to find a device handler". Everything looked like it was created correctly, but it did not work.
I had forgotten to set DTMF Signaling Method to RFC 2833.
By looking the error "Unable to find a device handler" what i believe is your CUCM IP address which is in the CVP Static Route is not listed in the CallManagerGroup which is assigned to your SIP Trunk's Device Pool.
What is the CUCM Version you use. If you are using 8.5 and above you can use the below feature
SIP Trunks Can Run on All Active Unified CM Nodes
When the Run on all Active Unified CM Nodes option is checked on a SIP trunk, Unified CM creates
an instance of the SIP trunk daemon on every call processing subscriber within the cluster, thus allowing
SIP trunk calls to be made or received on any call processing subscriber. (Prior to this feature, up to three
nodes could be selected per trunk by using Unified CM Groups.) With Run on all Active Unified CM
Nodes enabled, outbound SIP trunk calls originate from the same node on which the inbound call (for
example, from a phone or trunk) is received. As with all Unified CM SIP trunks, the SIP daemons
associated with the trunk will accept inbound calls only from end systems with IP addresses that are
defined in the trunk's destination address fields. Running SIP trunks on all nodes is recommended where
the SIP trunk is required to process a large number of calls so that outbound and inbound call distribution
can be evenly spread across all call processing subscribers within a cluster. Also, when multiple SIP
trunks to the same destination(s) are using the same subscriber, a unique incoming and destination port
number must be defined per trunk to allow each trunk to be identified uniquely.
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