UM Exchange integration question

[doctora]

We have Exchagne configured as voice mail.  When we try to transfer a call directly to voicemail the call drops as soon as the person doing the transfer hangs up or presses transfer the second time. Transfer works fine if we are not trying to go directly to voicemail.  I am thinking it has something to do with transfering it directly to sip trunks verses transfering it to another extension in the pbx.

Thanks
Mark

[v2win]

Can you post your SIP Peer Profile and software level?

[doctora]

We are on Release level 4.1 SP1
Active software load 10.1.1.11_2



Profile

SIP Peer Profile Label   Exchange   
Network Element   Exch-FQDN   
 
Local Account Information
  Registration User Name       
  Address Type   IP Address: 11.11.110.112   
 
 
Call Routing and Administration Options
  Interconnect Restriction   1   
  Maximum Simultaneous Calls   32   
  Outbound Proxy Server       
  SMDR Tag   0   
  Trunk Service   10   
  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       
  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   
 
 
SDP Options
  Allow Peer To Use Multiple Active M-Lines   Yes   
  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   Yes   
  Limit to one Offer/Answer per INVITE   No   
  NAT Keepalive   Yes   
  Prevent the Use of IP Address 0.0.0.0 in SDP Messages   Yes   
  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
  Session Timer   0   
  Allow Display Update   No   
  Build Contact Using Request URI Address   No   
  Disable Reliable Provisional Responses   No   
  Enable sending '+' for E.164 numbers   No   
  Force Max-Forward: 70 on Outgoing Calls   No   
  Ignore Incoming Loose Routing Indication   No   
  Use P-Asserted Identity Header   No   
  Use P-Preferred Identity Header   Yes   
  Use Restricted Character Set For Authentication   No   
  Use To Address in From Header on Outgoing Calls   No   
 
 
 

[v2win]

This is from the docs try this and see what happens


Allow Peer To Use Multiple Active M-Lines: Yes
Allow Using UPDATE For Early Media Renegotiation: Yes
Avoid Signaling Hold to the Peer: Yes
Enable Mitel Proprietary SDP: No
NAT Keepalive: Yes
Prevent the Use of IP Address 0.0.0.0 in SDP Messages: Yes
Disable Reliable Provisional Responses: No
Use P-Asserted Indentity Header: Yes

[doctora]

I am not allowed any changes until the weekend.  I will let you know. 

Thanks for the help so far.

MArk

[doctora]

The calls are still dropping when transfered to the sip trunk group.

thanks for the help
Mark

[brantn]

SIP Peer Profile Label   EXCH   
Network Element   EXCH   
 
Local Account Information
  Registration User Name       
  Address Type   FQDN: blank.blank.local 
 
 
Call Routing and Administration Options
  Interconnect Restriction   1   
  Maximum Simultaneous Calls   12   
  Outbound Proxy Server       
  SMDR Tag   0   
  Trunk Service   9   
  Zone   10   
  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       
  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   Yes   
  Force sending SDP in initial Invite message   No   
  Force sending SDP in initial Invite - Early Answer   Yes   
  Limit to one Offer/Answer per INVITE   No   
  NAT Keepalive   No   
  Prevent the Use of IP Address 0.0.0.0 in SDP Messages   Yes   
  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   No   
  Use P-Preferred Identity Header   No   
  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   0   
  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


If that doesn't work post COS also check diversion settings in system options.

[Chakara]

  I'm dying to know if this works.....

[doctora]

Sorry I was not watching this thread.  I will test ASAP and reply.  Hopefully tomorrow if I get the OK.  Otherwise over the weekend.

[brantn]

It is working at mutiple client sites and our own both on Exchange 2007 and 2010.

[doctora]

It is working.  Thanks.  If you can explain why this is working that would be great.  The sip peer profile is confusing to me.   If you do not have the time I understand and thank you again.

Mark

[brantn]

The main issue I believe lie in Use P-Preferred Identity Header No in your profile you had this set to yes and Disable Reliable Provisional Responses set to No

Disable Reliable Provisional Responses
 Select Yes to disable the use of reliable provisional responses (PRACK) on outgoing and incoming calls, unless the Required Header is received on incoming calls. Most Peers now support PRACK and this can be useful in interoperability scenarios with the PSTN (see RFC 3262). If the SIP Peer also supports PRACK, it is recommended that this option be set to No.
 
Use P-Preferred Identity Header
 If you enable this option, the system uses the Default CPN data to build the P-Preferred-Identity header. No display name is shown in this header. A P-Preferred-Identity will not be included in messaging if the