Mitel Forums - The Unofficial Source
Mitel Forums => Mitel MiVoice Business/MCD/3300 => Topic started by: doctora on March 24, 2011, 11:45:44 AM
-
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
-
Can you post your SIP Peer Profile and software level?
-
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
-
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
-
I am not allowed any changes until the weekend. I will let you know.
Thanks for the help so far.
MArk
-
The calls are still dropping when transfered to the sip trunk group.
thanks for the help
Mark
-
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.
-
I'm dying to know if this works.....
-
Sorry I was not watching this thread. I will test ASAP and reply. Hopefully tomorrow if I get the OK. Otherwise over the weekend.
-
It is working at mutiple client sites and our own both on Exchange 2007 and 2010.
-
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
-
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