Hi Art,

Thanks very much for your reply. Could you please port the SIP configuartion. How can I bind it to the interface please?

Your assistance will be highly appreciated.

Regards,

Mudassar

Mudassar,

See the link below:

http://www.cisco.com/en/US/docs/ios/voice/command/reference/vr_b1.html#wp1555923

In your situation you want to bind the protocol to whatever address the far-end device is trying to send to - in your case I think it would be your outside NAT address.

voice service voip

sip

bind all source-interface Loopback1

Art

Hello Art,

   Great! Many thanks for your assistance. I will check it and will post the outcome.

Regards,

Mudassar

Hi Art,

I've associated external serail 0/3/0 interface with SIP, but still not getting through. the invite message contains the IP address of the ISP side after @ sign, which seems like rejected by the router because it is not its IP (it seems logical). I am try to get debug outputs , but could you please verify which IP should be in the invite message of SIP negociation?

regards,

Mudassar

Hi Mudassar,

It would be very helpful if you would please post the entire error you are seeing and also the debug ccsip message output.  I am hesitant to try and answer any further without seeing more information to try and make a call.

Thanks,

Art

Hello Art,

I trying to get debug results from CPE (Cisco 2811), meanwhile could you please check the debug out put from taken from SIP proxy server. Attached are the packets sequence when we spoof an INVITE to them and put their IP 82.71.194.25 as the realm in the INVITE line and in the To: line instead of 212.23.7.228. They accept this and send us a 100 Trying, then a 180 Ringing and after a bit a 500 Internal Server error (probably because its facked).

Background:

We spoof sending from src IP 212.23.7.228 to them, we are not really sending from 212.23.7.228.

Packet sequence:

INVITE_82.71.194.25.txt (Spoof src 212.23.7.228-> 82.71.194.25) 100_Trying.txt 180_Ringing.txt 500_Internal_Server_Eror.txt.

Many thanks,

Mudassar

Hi Art,

Please find the attached debug output.

Many thanks for your help.

Regards,

Mudassar

Muddassar,

After reviewing the SIP debug it appears that the call gets all the way to "rininging" before failing.  That means it is beyond the handshake and is very close to connecting, whiuch also means H.323 is involved by now.  Try checking the "Media Termination Required" checkbox on the H.323 gateway in CCM.

Thanks,

Art

Additionallly, if requiring the MTP does not work input the following in the gateway:

voice service voip 

cause-code legacy 

and after a call perform a:

show call history voice

you should see a couple lines that look like this:

DisconnectCause=10  
DisconnectText=normal call clearing

From

:

http://www.cisco.com/en/US/docs/ios/12_2t/12_2t11/feature/guide/ftbibble.html#wp1025089

Please forward the cause code.

thanks,

Art

Hi Art,

Thanks very much for your assistance. I am really impressed!

We have fixed this with your assistance. Yes it was H323 signalling misconfiguration of UCM which is now fixed by the customer. 

Could you please advice us in one more issue, As I have mentioned earlier, that yesterday we were getting Error 400 (Bad Request - Invalid host) and we have changed this in the SIP address command on our proxy server (......@x.x.x.x) in the INVITE line and To line with the destination IP of the customer's SIP gateway to fix this issues. But we want to change it back to our normal configuration, which is our proxy IP address in the INVITES message.

Could you please give me an idea that what is the cause of this error and what I can suggest to the customer to fix this issue?

your assistance would be highly appreciated.

Regards,

Mudassar

Mudassar,

I'm don't fully understand what you need, but please do the following:

Change the configuration to what you want it to look like

Run another debug ccsip

Copy the output and I will take a look at it.

Thanks,

Art