voice register dn shared line doesnt work with Cisco 7861 sip phones

[wrobynson]

Hi

We want to share the same extension number says 1010 to two sip phones 7861. But it looks like the two phones can not ring at the same time. The calls go to one phone and sometimes to other. 

ATTENTION: THIS HAPPENS ONLY WITH CALLS ORIGINATED FROM OTHER NETWORKS (VIA DIAL-PEER). WITH INTERNAL CALLS OR COMING AT THE E1 TRUNK, SHARED LINE WORKS FINE!

Below is my configuration. 

voice register dn  1
 number 1010
 allow watch
 shared-line max-calls 5
 mwi
!
voice register pool  1
 busy-trigger-per-button 2
 id mac ...
 session-transport tcp
 type 7861
 number 1 dn 1
 dtmf-relay rtp-nte sip-notify
 username ...
 codec g711ulaw
!
voice register pool  2
 lpcor type local
 lpcor incoming PSTNTrunk
 busy-trigger-per-button 2
 id mac ...
 session-transport tcp
 type 7861
 number 1 dn 1
 dtmf-relay rtp-nte sip-notify
 username ...
 codec g711ulaw

Thank's for help.

walter@robynson.com

[Vaijanath Sonvane]

Hi,

When call is originated from other network to 1010, which registrar pool phone rings? I see class or restriction configured on registrar pool 2. Have you tried removing it and then test?

[wrobynson]

Hi.

The calls go to one phone and sometimes to other.

I removed the cor configuration and test. The problem continues.

[wrobynson]

More config...

voice service voip
 no ip address trusted authenticate
 callmonitor
 no callmonitor trace
 rtp-port range 20000 30000
 dtmf-interworking rtp-nte
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 supplementary-service h450.12 advertise-only
 no supplementary-service sip moved-temporarily
 no supplementary-service sip refer
 redirect ip2ip
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 modem passthrough nse payload-type 101 codec g711ulaw redundancy maximum-sessions 1
 h323
 sip
  bind control source-interface GigabitEthernet0/1
  bind media source-interface GigabitEthernet0/1
  registrar server
  sip-profiles 1
  no call service stop
!