Ougoing CLI from incomg SIP trunks.

[ASCSTech]

Guys,
having issues trying to get outgoing CLi correct when calls are coming in from our SIP server. I have added a correct CPN number into the SIP peer profile but still doesn't display. My main CPN substitution form is correct as other numbers work fine. When i look at my SMDR the call is coming in from X999. i have used a tag for smdr and this shows in the SMDR for the outgoing call. Any pointers? see SIP profile and SMDR trace below.

11/29 02:14P 00:00:04 T940 006 00870029919 T100
1
11/29 02:14P 00:00:04 X999 0870029919 A T100


Call Routing and Administration Options
Interconnect Restriction 1
Maximum Simultaneous Calls 5
Outbound Proxy Server SLIPPERPy
SMDR Tag 940
Trunk Service 5
Zone 1
Alternate Destination Domain Enabled No
Alternate Destination Domain FQDN or IP Address
Enable Special Re-invite Collision Handling No
Private SIP Trunk No
Route Call Using To Header No


Calling Line ID Options
Default CPN 0870718940
CPN Restriction No
Public Calling Party Number Passthrough No
Use Diverting Party Number as Calling Party Number No


Authentication Options
User Name
Password *******
Confirm Password *******
Authentication Option for Incoming Calls No Authentication
Subscription User Name
Subscription Password *******
Subscription Confirm Password *******


SDP Options
Allow Peer To Use Multiple Active M-Lines No
Allow Using UPDATE For Early Media Renegotiation No
Avoid Signaling Hold to the Peer No
Enable Mitel Proprietary SDP No
Force sending SDP in initial Invite message No
Force sending SDP in initial Invite - Early Answer No
Limit to one Offer/Answer per INVITE No
NAT Keepalive No
Prevent the Use of IP Address 0.0.0.0 in SDP Messages No
Renegotiate SDP To Enforce Symmetric Codec No
Repeat SDP Answer If Duplicate Offer Is Received No
RTP Packetization Rate Override No
RTP Packetization Rate 20ms
Special handling of Offers in 2XX responses (INVITE) No
Suppress Use of SDP Inactive Media Streams No


Signaling and Header Manipulation Options
Allow Display Update No
Build Contact Using Request URI Address No
De-register Using Contact Address not * No
Disable Reliable Provisional Responses Yes
Disable Use of User-Agent and Server Headers No
Enable sending '+' for E.164 numbers No
Force Max-Forward: 70 on Outgoing Calls No
Ignore Incoming Loose Routing Indication No
Only use SDP to decide 180 or 183 No
Require Reliable Provisional Responses on Outgoing Calls No
Use P-Asserted Identity Header Yes
Use P-Preferred Identity Header Yes
Use Restricted Character Set For Authentication No
Use To Address in From Header on Outgoing Calls No
Use user=phone No


Timers
Registration Period 3600
Registration Period Refresh (%) 50
Session Timer 90
Subscription Period 3600
Subscription Period Minimum 300
Subscription Period Refresh (%) 80


Key Press Event Options
Allow Inc Subscriptions for Local Digit Monitoring No
Allow Out Subscriptions for Remote Digit Monitoring No
Force Out Subscriptions for Remote Digit Monitoring No
Request Outbound Proxy to Handle Out Subscriptions No
KPML Transport default
KPML Port 0

 

[Supernova99]

What SIP trunk provider are you using?
Have you checked the tested interop docs available on MoL?


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[ASCSTech]

We have a customer SIP server co located in our premises so they are just point-point between the MITEL and SIP server. i have asked the SIP server provider for info on what digits they are sending but awaiting response.
also i dont think it is a SIP only issue as i have MITEL IP trunks running to another site and the same was happening with this until i sorted the o/g cli fromn the remote site.
my issue is that i cannot force the MITEL to send a CLI from the SIP Peer profile which is what i wold have thoughti could do

 

[Supernova99]

Has the SIP server been through Mitel interop testing?
The problem is that if it has not been submitted for testing and has passed it will be a lot of guesswork as to what the issue is.


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