Author Topic: Outgoing CLI from incoming Sip trunks  (Read 4871 times)

Offline ASCStech

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Outgoing CLI from incoming Sip trunks
« on: November 28, 2011, 10:46:12 PM »
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   
   


Offline ronniez

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Re: Outgoing CLI from incoming Sip trunks
« Reply #1 on: March 13, 2013, 06:00:25 AM »
Did you already have a solution for this?
I have the same problem.


 

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