01-12-2009 03:08 AM - edited 03-15-2019 03:27 PM
Hi all,
I have a strange problem using a ccme versione 4.1(0) on a router 2800 IOS: c2800nm-advipservicesk9-mz.124-15.T5
The CCME is correctly registered on a sip provider.
Some times when I have an incoming call i have the Ring "Free" but after 3 ring the call goes down "in busy state" but the phones do no ring.
IF i have a look at the "sip-ua calls" i have a situation as:
SIP UAC CALL INFO
Number of SIP User Agent Client(UAC) calls: 0
SIP UAS CALL INFO
Call 1
SIP Call ID : 04ca778d7b3c764b6791b9857f91dbe0@194.244.11.18
State of the call : SIP_STATE_OPTIONS_WAIT (27)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : kebuIPPBX
Called Number : 41484
Bit Flags : 0x40000C 0x104 0x0
CC Call ID : 199051
Source IP Address (Sig ): 81.174.35.83
Destn SIP Req Addr:Port : 0.0.0.0:0
Destn SIP Resp Addr:Port: 194.244.11.18:5060
Destination Name : 194.244.11.18
Number of Media Streams : 1
Number of Active Streams: 0
RTP Fork Object : 0x0
Media Mode : flow-through
Media Stream 1
State of the stream : STREAM_IDLE
Stream Call ID : -1
Stream Type : voice+dtmf (1)
Negotiated Codec : No Codec (0 bytes)
Codec Payload Type : 255 (None)
Negotiated Dtmf-relay : inband-voice
Dtmf-relay Payload Type : 0
Media Source IP Addr:Port: 81.174.35.83:0
Media Dest IP Addr:Port : 0.0.0.0:0
Orig Media Dest IP Addr:Port : 0.0.0.0:0
Options-Ping ENABLED:NO ACTIVE:NO
Number of SIP User Agent Server(UAS) calls: 1
I don't know why. My opinion is a software version problem.
Thanks
01-12-2009 07:13 AM
You'd want to look at 'debug ccsip messages' for the failed case. It's possible it's not even getting to your router. If it is, it could be a call routing problem.
hth,
nick
01-12-2009 11:14 AM
Thanks Grate debug. I make 3 calls but no error tomorrow I will try again to log a wrong call.
What do you mean call routing problem?
Thanks
FX
01-13-2009 02:46 AM
Hi, I add a new information:
Lokking at the logs with sip provider, the ccme refuses the call and the error in the sip provider is "busy congestion".
Any ideas.
01-13-2009 05:43 AM
You would need to check your dial peer configuration against the incoming sip messaging. This means debug voip dial peer, debug ccsip messages
You would want to use this command:
show dial-peer voip summary
hth,
nick
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