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CCME SIP provider Problem

fabriziop
Level 1
Level 1

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

4 Replies 4

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

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

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.

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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