disconnect cause 57 in ccsip inout and disconnect cause 403 forbidden in sip messages

[fernandogarciat]

Hi everyone.

Every time a place an outside call i received the messages

Disconnect Cause (CC)    : 57
Disconnect Cause (SIP)   : 403

the flow of the call is:

____________________________________________

 3001                               92144440

IP PHONE ---- CME ---- TISP (through a sip trunk)

____________________________________________

attached is the config and debugs

I hope you can help me.

[Vivek Batra]

Your provider is rejecting INVITE with cause 403 Forbidden (Not Allowed) and most probable reason for such instances are sending incorrect ANI, DNIS not with appropriate prefix etc, service is not active etc. You probably need to check with your provider for root cause.

- Vivek

[Dennis Mink]

also they might require you to send a domain name in the invite, instead of an IP address

[fernandogarciat]

Hi Dennis.

service provider told me the only parameters I need are; proxy IP address and my own IP address, and thats it. that's because is weird, the configuration is supposed to be simple and plain.

I will check with service provider.

but further ideas will be appreciate

[Dennis Mink]

how about inbound calls? do they work?  if so, what number is presented and does that match with what ANI your are sending outbound?

[fernandogarciat]

Hi thanks everyone for your advice.

As an update, the outgoing calls are working now, I had to change the calling number from extension to DID provided by ITSP (through a translation rule), all kind of outgoing calls are working fine now.

But I still have no luck with incoming calls, when I debug during a test incoming call  nothing happens, I was reading that under sip-ua, the "credentials username" command is the one that handles incoming calls but I still not test this part.

Any ideas?

[Vivek Batra]

Thanks for the updates.

That all depends on whether your SIP trunk provider is giving you peer to peer services where they must have asked you for static IP where you will get all your incoming calls OR your provider is expecting SIP trunk to get registered with it which shall enable provider to route incoming calls on that SIP trunk.

If your SIP trunk service is peer to peer and you have been provided static IP to service provider, you should check with them where the issue is, else you are right you need to have an appropriate configuration to get SIP trunk registered with provider. You must have got authentication credentials from provider to be configured in gateway.

- Vivek

[fernandogarciat]

Hi Vivek.

You are right and in this case is a peer to peer service, they gave me a private static IP address.
I have configured a softphone with the info provided and all worked fine either incoming and outgoing calls, I'm going to take some wireshark captures and try to figure out what the problem is.

fg

[Vivek Batra]

Hi,

they gave me a private static IP address.

My point was whether the have asked you for your (/gateway) IP address or not. If it is working in softphone and most of the third party SIP softphone sends REGISTER by default, I think you have registration based SIP trunk, instead of peer to peer.

Anyway do share the results with us and let us know if we can help there in any way.

- Vivek

[fernandogarciat]

I will Vivek, thanks a lot.