Feature rejected after SIP call setup

[azhar]

Hello,

I have configured an external SIP gateway and it has registered OK with the SIP provider's system (sipextgw -l shows 'registered').

when I seize the ARS Prof Trunk gp and dial a number, after a few seconds, I get 'feature rejected'.

could anyone have a look at the attached log file and tell me why I get 'Feature Rejected' on the phone display after that all the invite messages have been sent. It says that the reason for termination is because Timer B fired but I cannot find Timer B anywhere.

Phone system is an OXE R9.0 h1.301.28.

Any advice?

regards
Azhar.

[Ralonso]

In the traces i only see sent to network but nothing recive from de provider.
Check route configuration.

[azhar]

Hello

I can ping the external gateway OK, which I think rules out the route issue. Do you agree?

regards,
Az

[Ralonso]

try to capture traces when the gateway register in a provider and post it.
and post ext.gateway configuration.

[azhar]

Hello,

there is no registration as the remote gateway does not support REGISTER. It is a pure trunking service where the SIP device should forward any INVITE requests to.

I have attached some more logs: between i1 and i2 and i2 and the sip motor.

Here is the ext gateway config:

┌─Review/Modify: SIP Ext Gateway───────────────────────────────────────────────────────────┐
│ │
│ Node Number (reserved) : 1 │
│ Instance (reserved) : 1 │
│ SIP External Gateway ID : 2 │
│ │
│ Gateway Name : IP Office │
│ SIP Remote domain : 212.11.94.100 │
│ PCS IP address : ----------------------------------------------- │
│ SIP Port Number : 5060 │
│ SIP Transport Type + UDP │
│ RFC3262 Forced use + False │
│ Belonging Domain : -------------------------------------------------- │
│ Registration ID : -------------------------------------------------- │
│ Registration ID in P_Asserted + False │
│ Registration timer : 0 │
│ SIP Outbound Proxy : -------------------------------------------------- │
│ Supervision timer : 0 │
│ Trunk group number : 12 │
│ Pool Number : -1 │
│ Outgoing realm : -------------------------------------------------- │
│ Outgoing username : -------------------------------------------------- │
│ │
│ Outgoing Password : ---------- │
│ Confirm : ---------- │
│ │
│ Incoming username : -------------------------------------------------- │
│ │
│ Incoming Password : ---------- │
│ Confirm : ---------- │
│ │
│ RFC 3325 supported by the distant + True │
│ DNS type + DNS A │
│ SIP DNS1 IP Address : ----------------------------------------------- │
│ SIP DNS2 IP Address : ----------------------------------------------- │
│ SDP in 18x + True │
│ │
└──────────────────────────────────────────────────────────────────────────────────────────┘

[sadim]

Hi,

Put "Supervision timer"=5, wait 30sec and check with "sipextgw -l" if the sip gw goes out of service.
Also check with motortrace if the external gw respondes to the oxe sip options messages.

Regards
Sadim

[azhar]

Hello

When supervision timer is set to 5, the gateway immediately goes out of service with 'sipextgw -l'.

what does this mean then?

cheers

Az

[sadim]

Hi,

It means that your SIP provider is deaf.

Sadim

[azhar]

He might surely be! But he is shouting out loud that he is listening, but cannot hear anything!, i.e. he cannot see any INVITE messages.

Another question: why I am getting 'Feature Rejected' on the screen and 'Please make enquiry' voice guide; is there something blocking the call on the OXE itself?... as this is when you get voice prompts from the OXE. Otherwise, I would have got 'busy' or not obtainable tone if it had got out to the network, do you think?

regards

Az

[azhar]

Hello all

managed to get the packets going out. i had to configure the '/etc/resolv.conf'. Putting the hostname in the OXE's hostname file did not suffice.
Now I have got another problem. The provider needs the OXE to send packets using the public IP as opposed to the internal IP of the OXE. Consider the example below:


U 2009/10/23 11:51:08.721997 93.152.83.1:58505 -> v-sip-trunk-out-f1 [212.11.94.100]:5060 INVITE sip:01225445186@212.11.94.100 SIP/2.0. <<my public IPAllow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, SUBSCRIBE, OPTIONS, UPDATE, INFO.
Supported: replaces,timer,100rel.
User-Agent: OmniPCX Enterprise R9.0 h1.301.28.
Session-Expires: 1800;refresher=uac.
Min-SE: 900.
P-Asserted-Identity: "Azhar" <sip:514@192.168.123.152;user=phone>. <<<<He is saying that this should be my public WAN IP
Content-Type: application/sdp.
To: sip:01225445186@212.11.94.100.
From: "Azhar"
<sip:514@192.168.123.152>;tag=f48a4e8495412302ceafd324487e1eab.
Contact: <sip:192.168.123.152;transport=UDP>. <<<<He is saying that this should be my public WAN IP
Call-ID: 3fd51dbcdf44d43c80522b4fca49e717@192.168.123.152. <<<<He is saying that this should be my public WAN IP
CSeq: 243973813 INVITE.
Via: SIP/2.0/UDP
192.168.123.152;branch=z9hG4bKae23854d76502a87d097a06b9c13904f. <<<<He is saying that this should be my public WAN IP
Max-Forwards: 70.


Can I modify something in my programming so that the OXE sends its request using the public IP?
What I would do then is that any reply on port 5060 would be NATed to the OXE.

cheers

Az