sip profile diversion header reason modification reason=unconditional to reason=deflection

[Ian Meyers]

I'm attempting to modify the diversion header when calling one specific number.  I want to modify the reason=unconditional to reason=deflection.  I've made a single peer that I confirmed is matched, and assigned a sip profile to the outgoing peer which I also confirmed I saw matched in the debugs.  It doesn't appear to be matching the sip profile or at least it definitely isn't modifying the header like I'd want.  The call flow is on prem -> first number on RP -> forward all to second number -> CUBE -> SIP trunk. Currently seeing reason=unconditional.   I either need help understanding the sip diversion header matching/modification, and/or a better explanation for when a call gets set as unconditional versus deflection. 

config added I've tried both add and modify... but this is when I tried modify

voice class sip-profiles 2
 request INVITE sip-header Diversion modify "<sip:9123456789@>;privacy=off;reason=unconditional;screen=yes" "<sip:9123456789@>;privacy=off;reason=deflection;screen=yes"

dial-peer voice 702 voip
 preference 1
 voice-class sip profiles 2
 destination-pattern ^9123456789$
 session protocol sipv2
 session target ipv4:<cleaned>
 voice-class sip early-offer forced
 voice-class sip options-keepalive
 voice-class sip bind control source-interface Loopback0
 voice-class sip bind media source-interface Loopback0
 dtmf-relay rtp-nte
 codec g711ulaw
 fax protocol pass-through g711ulaw
 no vad

no shut

Verified hit external peer I configured

>>>>CCAPI handed cid 1721231 with tag 30 to app "_ManagedAppProcess_Default"
060250: Jun 16 14:18:06.832: //1721231/9C3288800006/CCAPI/ccCallProceeding:
   Progress Indication=NULL(0)
060251: Jun 16 14:18:06.832: //1721231/9C3288800006/CCAPI/ccCallSetupRequest:
   Destination=, Calling IE Present=TRUE, Mode=0,
   Outgoing Dial-peer=702, Params=0x13A9B248, Progress Indication=NULL(0)

 Verified the profile was matched in dial peer 702 too

070123: Jun 16 15:37:58.490: //-1/xxxxxxxxxxxx/SIP/Info/info/64/sipSPISetSipProfilesTag: voice class SIP Profiles tag is set : 2
070124: Jun 16 15:37:58.490: //1722071/C473B4800006/SIP/Info/notify/8192/sipSPISetOverlapConfiguration: Overlap signaling: FALSE: Endpt: SIP Trunk
070125: Jun 16 15:37:58.490: //1722071/C473B4800006/SIP/Info/verbose/10240/sipSPI_ipip_GetHdrPassthruCfg: Hdr passthrough config:2 tag:0
070126: Jun 16 15:37:58.490: //1722071/C473B4800006/SIP/Info/verbose/2048/sipSPI_ipip_GetCopyListCfg: Copy-list config:2 tag:0
070127: Jun 16 15:37:58.490: //1722071/C473B4800006/SIP/Info/info/2048/sipSPIGetExtensionCfg: SIP extension config:1, check sys cfg:1
070128: Jun 16 15:37:58.490: //1722071/C473B4800006/SIP/Error/sipSPI_ipip_build_consolidated_header_list:
 No headers associated with passthrulist tag: 0 and copylist tag: 0

[R0g22]

Question : Why do you want this to be set to deflection ?

[Ian Meyers]

We believe this is causing problems with our carrier and want to prove it out.  We eventually would want to change it the other way too, or at least understand better the circumstances around this message.

[R0g22]

It should not cause issues. Can you share the INVITE with the diversion header or the complete log if there is a call failure being reported for hairpinned calls ?
The diversion header needs to contain the DID number that your ITSP has assigned to your trunk(s). Is the "9123456789" number a DID for your trunk ?

[gmgarrian]

Didn't see your response...I need to scroll down more.

The SIP profile should work per testing with the test tool:  https://cway.cisco.com/tools/SipProfileTest/

[gmgarrian]

Can you post the initial INVITE message?

[Anup Swain]

When you send to PSTN gateway. DIVERSION heder since you have the user@host part. You also need to add user=phone. Which I see is not there. Maybe thats the format your PSTN expects. May help