cme sip-phone consult transfer not changing caller-id

[alig.norbert]

Hi all,

I'm facing the following problem:

When SIP-PhoneA transfer an incoming call (internal/external) to SIP-PhoneB through consult-transfer,
PhoneB sees the PhoneA internal number and not the origin-caller-id when connected with the origin-caller.

CME:      12.0

Phone: 7821, 12-0-1-11

Any ideas? Bug on CME12.0? Got the same problem with 78xx version 10.3.

Thanks,

Norbert

Parcial config:

------------

voice service voip
 ip address trusted list
  ipv4 192.168.0.0 255.255.0.0
  ipv4 10.0.0.0 255.255.255.128
 media flow-around
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 no supplementary-service h450.2
 no supplementary-service h450.3
 no supplementary-service h450.7
 no supplementary-service sip moved-temporarily
 no supplementary-service sip refer
 supplementary-service ringback h225-info
 redirect ip2ip
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 no fax-relay sg3-to-g3
 fax-relay ans-disable
 modem passthrough nse codec g711alaw
 h323
  h245 caps mode restricted
 sip
  bind control source-interface GigabitEthernet0/0.115
  bind media source-interface GigabitEthernet0/0.115
  registrar server expires max 600 min 60
  asserted-id ppi

!

!

voice register global
 mode cme
 source-address 192.168.115.1 port 5060
 no privacy
 timeouts interdigit 4
 max-dn 200
 max-pool 42
 load 7821 sip78xx.12-0-1-11
 load 8861 sip88xx.12-0-1SR1-1
 authenticate register
 authenticate realm all
 timezone 23
 time-format 24
 date-format D/M/Y
 hold-alert
 mwi stutter
 mwi reg-e164
 call-forward system redirecting-expanded
 voicemail 999
 tftp-path flash:
 file text
 create profile sync 0003253876683702
 network-locale CH
 user-locale DE load flash:/CME-locale-de_DE-German-12.0.12.0.tar
 ntp-server 192.168.115.1 mode directedbroadcast
 conference hardware

!

voice register dn  95
 translation-profile incoming TP_200
 number 295
 call-forward b2bua all 999
 call-forward b2bua noan 252 timeout 20
 call-forward b2bua unregistered 252
 allow watch
 pickup-call any-group
 pickup-group 1
 name Sitxxx
 huntstop channel 1
 label Sitxxx(295)
 mwi

!

voice register template  1
 button-layout 1-2 line
 button-layout 24-32 speed-dial
 button-layout 6-23 blf-speed-dial
 button-layout 3-5 feature-button
 softkeys connected  Hold Park Endcall
!
voice register template  2
 button-layout 1 line
 button-layout 3-23 blf-speed-dial
 button-layout 2 feature-button

!

!
voice register dn  4
 translation-profile incoming TP_200
 number 354
 call-forward b2bua noan 352 timeout 20
 call-forward b2bua unregistered 352
 allow watch
 pickup-call any-group
 pickup-group 2
 name Sitz
 huntstop channel 1
 label Sitz(354)
 mwi
!

voice register pool  8
 registration-timer max 120 min 60
 busy-trigger-per-button 2
 id mac CC98.914E.1EB9
 feature-button 1 PickUp
 type 7821
 number 1 dn 95
 template 2
 cor incoming COR-GEM default
 presence call-list
 username 295 password xxxx
 no call-waiting
 description 0xxxxxxxxx
 codec g711alaw
 no vad
 after-hour exempt

!

voice register pool  13
 registration-timer max 120 min 60
 busy-trigger-per-button 2
 id mac 50F7.222C.B929
 feature-button 1 PickUp
 type 7821
 number 1 dn 4
 template 2
 cor incoming COR-GRUND default
 presence call-list
 dtmf-relay rtp-nte
 username 354 password xxx
 no call-waiting
 description xxxx
 codec g711alaw
 no vad
 after-hour exempt

[R0g22]

Attach a complete "show run" please. Also take a log for -

debug ccsip message
debug voice ccapi inout
debug voip translation

[alig.norbert]

Here the config

[alig.norbert]

The debug.

Thanks for your help.

[R0g22]

Can you enable all the debugs at once and grab them in a single file please ?

[R0g22]

Calling, called and transfer numbers please ?

[alig.norbert]

Sorry

354 -> 252 transfer -> 295

[R0g22]

Thanks. I see you already have "calling-number initiator" configured under telephony-service.
If you do a blind transfer, does the caller ID show correctly ?

[alig.norbert]

I wasn‘t able to get the blind transfer working, changed it under telephony-service but still consult.

CFA is working properly.

Is the transcoding an issue, b‘cause internal conferencing (3phones) doesn‘t work either?

[R0g22]

This has nothing to do with transcoding. You can set consult system to full-blind. With blind it should work correctly though.

Just to re-iterate, A calls B. B picks up and calls C. Does C pick up the call before B completes the transfer or B transfers the call as soon as C rings and has not picked up the call necessarily ?

[alig.norbert]

Hi,
Here the flow:
A calls B. B picks up, press „transfer“ and calls C.
C pick up the call and talk with B, B then completes the transfer by pressing „transfer“.
On the display C, it shows the caller-id from B, but should be A.

Thanks

[R0g22]

Ok. Out of curiosity I tested this in the lab with CME 12.0 on ISR-4k since I have seen quite a number of posts lately for this and it indeed works fine.
You would need to take a complete debug. The debug that you shared previously is not complete.

Just take a "debug ccsip all" and nothing else. Please ensure you take this either off production hours or when there are no calls. This is a CPU intensive debug, very verbose.

[alig.norbert]

Thanks for checking.
I’ll collect the logs.

May you provide your testing config, so that I could check where the config issue is?

Kind regards,
Norbert

[R0g22 · Accepted solution]

Before you provide the logs, can you remove "asserted-id ppi" command from voice service voip/sip configuration and then test. If it still does not work, please send the logs.