08-01-2012 12:11 PM - edited 03-16-2019 12:29 PM
Hello,
I have a UC520 that connected to a SIP trunk through Level 3. I am able to do inbound calls, but unable to do outgoing calls. The issue is some kind of routing issue, but unable to get it to work. Here is a "debug ccsip call" log calling outbound:
The Call Setup Information is:
Call Control Block (CCB) : 0x88BEB698
State of The Call : STATE_DEAD
TCP Sockets Used : YES
Calling Number : 763273xxxx
Called Number : 1612812xxxx
Source IP Address (Sig ): 0.0.0.0
Destn SIP Req Addr:Port :
Destn SIP Resp Addr:Port :
Destination Name :
000427: //611/4C4B003E8290/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 3
Disconnect Cause (SIP) : 200
Notice the Source IP Address (Sig ): 0.0.0.0
Any ideas? Config is attached
08-01-2012 11:26 PM
I don't see that you are inspecting SIP outbound?
08-01-2012 11:33 PM
hi Marc,
There could be a mismatch in the transport protocol. You can change it under the dial-peers ( tcp or udp)
Hope this helps!
Thanks,
Karthik
08-01-2012 11:51 PM
Wouldn't this prevent the unit from registering at all, and therefore inbound calls would not work?
08-02-2012 03:10 AM
Disconnect cause 3 means no route to destination. You might want to check your dial-peers..
Check the configured dial-peers to your provider. and check routing to the ip addresses
You can send a debug voip ccapi inout
and debug ccsip messages
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08-02-2012 09:23 AM
Well I tried applying the "session transport udp" commands to almost all dial peers but is still not allowing a call through.
I believe that the dial peers may be fine? Reason I say that is in the Cisco Configuration Assisstant Troubleshooting section, I can do a Dial Plan test for outgoing and it says that the call is Allowed and shows the Sip Server IP which is Level 3. Would this mean my dial peers are correct, and it is a routing issue?
Ive attached a debug voip ccapi inout log of the outbound call.
08-02-2012 02:32 PM
Hi Marc,
I cheked the logs and is still cause value 3. Perhaps the dial-peer is down! Can you check the "show dial-peer voice summ" and also "show sip-ua reg status". What happend if you ping the Provider sourced from FA 0/0?
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