I'm trying to set-up a click to dial service using JTAPI, I don't know where this problem is coming from as I'm new to CUCM can anyone tell me what the problem might be? Here's the error I got from the servlet log:
SIP/2.0 503 Service Unavailable
Date: Wed, 30 Sep 2009 19:59:36 GMT
Warning: 399 "Unable to find a device handler for the request received on port 5060 from 10.36.134.137"
Via: SIP/2.0/TCP 10.36.134.137:5060
CSeq: 1 REFER
I'd appreciate any help.
For information, i had the same error because the domain on the invite was not configured in the Call Manager / Entreprise Parameters > Organization Top Level Domain & Cluster Fully Qualified Domain Name
If it can help you
Am facing the same issue. Unable to trace the root cause. Can someone help to trace the problem.
Am having contact center environment where Inbound calls to Agents coming from PSTN are getting into RESERVED state on Cisco Finesse desktop. After tracing, found SIP 503, Service Unavailable on CUCM and VG logs. Also, I see error "Unable to find a device handler for the request received on port 8317 from 192.168.4.12 (CVP)" on CUCM Logs.
My lab environment contains ICM Sprawler, CVP, Finesse and VG with 1E1 interface.
On ICM, call hits and remains in Queue to SG Node. Call doesnot move from Success or failure node.
Attached CUCM and ICM logs. VG Config attached reference.
Thanks for taking above too read. Issue is been resolved. In contact center scenario, SIP trunk between CVP and CUCM was missing.
I had the same issue as you.
error 503 with warning like this :
Warning: 399 Server name "Unable to find a device handler for the request received on port
The issue come from on my gateway i have "server-group" with IP of sub pair.
But on my CUCM SIP trunkl, i have DEVICE POOL with other SUB pair for manage call of this trink.
When i update server-group with right ip adress. every wsorking good !!!
This occurs when the source IP address that is sending the SIP INVITE does not exist as a SIP trunk in CUCM. You may have never created it, or the IP address may not be the one that is in use. Check bind commands for gateways, etc.
Thanks for the advice I set up a SIP trunk for my source ip and I got one phone to ring, when I send the click to dial request.
However I'm trying to get one phone to ring the other phone but I've had no luck so far. Is that still a SIP trunk issue?
This was exactly my issue. I have dual SIP gateways in an HSRP group connected to a third party IVR. When the IVR sends a refer message to move a call from queue to an available agent the CUBE invites the CUCM via the SIP trunk. I had the trunk configured with the VIP address instead of the actual box address of the active router. So I was receiving the same error message.
Based on Nicholas' message above I decided to add the actual active router and standby to the SIP trunk CUCM side, and now the error is resolved. Great job Nick!
Just in media gateway i did in
voice service voip
bind all source-interface Loopback0
where the loopback0 had the ip source of trunk.
with debug ccsip message
you can watch the ip local of sip trunk, and the dial-peer you put the ip remote
You can experience this problem if the only IP of CCM you are pointing to in the SIP dialpeer is not the primary CCM in the call manager group for the device pool you have applied to the SIP trunk.
Ensure that either the CCM you are pointing the dialpeer to is primary in the CCM group or you have multiple dial-peers with different preference as per the CCM group on the other side.
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