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1014
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Helpful
5
Replies

Problem with on hold in CME SIP phones

wrobynson
Level 1
Level 1

Hello all.

I am experiencing problems with hold calls. 

This is the scenario.

 
 
 

scenario.jpg

Case 1: extension 1010 (SCCP) calls 1012 (SIP)

When 1010 calls to 1012:
   2.1 - All devices (A, B and C) get ring.
   2.2 - answers.
   2.3 - But the call is for C.
   2.4 - So, A put the call on hold.
   2.5 - In the all phones (A, B and C), the line led will blink.
   2.6 - So, C pick up the call and talk to calling number (1010).
   2.7 - That is, it works perfectly here.
   Obs.: It's OK too if the call is originated from PSTN.

 

Case 2: extension 1011 (SIP) calls 1012 (SIP)

When 1011 calls to 1012:
   2.1 - All devices (A, B and C) get ring.
   2.2 - answers.
   2.3 - But the call is for C.
   2.4 - So, put the call on hold.
   2.5 - In the all phones (A, B and C), the line led will blink.
   2.6 - So, (TRY) pick up the call and the confusion begins!
   2.7 - When C presses the line button to pick up the call, 
         he makes a new call to the calling number (1011).
   2.8 - Then the caller who is on hold receives a new call...
         The mess is ready!

 

CONFIG:

ephone-dn 1
  number 1010
!
ephone 1
  device-security-mode none
  mac-address ...
  codec g729r8
  type 7931
  button 1:1
!
voice register dn 1 
  number 1011
  allow watch
  shared-line
  mwi
!
voice register dn 2 
  number 1012
  allow watch
  shared-line
  mwi
!
voice register pool 1
  busy-trigger-per-button 2
  id mac ...
  type 7861
  number 1 dn 1
  dtmf-relay rtp-nte
  username ...
  codec g711ulaw
!
voice register pool 2
  busy-trigger-per-button 2
  id mac ...
  type 7861
  number 1 dn 2
  dtmf-relay rtp-nte
  username ...
  codec g711ulaw
!
voice register pool 3
  busy-trigger-per-button 2
  id mac ...
  type 7861
  number 1 dn 2
  dtmf-relay rtp-nte
  username ...
  codec g711ulaw
!
voice register pool 4
  busy-trigger-per-button 2
  id mac 7035.0003.1012
  type 7861
  number 1 dn 2
  dtmf-relay rtp-nte
  username ...
  codec g711ulaw
!
voice register global
 mode cme
 source-address ... port 5060
 timeouts interdigit 2
 authenticate register
 authenticate realm all
 tftp-path flash:
 file text
!
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
 no supplementary-service sip handle-replaces
 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 ...
  bind media source-interface ...
  registrar server
  sip-profiles 1
  no call service stop
!
telephony-service
 no auto-reg-ephone
 ip source-address ... port 2000
 calling-number local
 service phone paramEdibility 1
 service directed-pickup gpickup
 service dss
 timeouts interdigit 2
 cnf-file location flash:
 cnf-file perphone
 load 7931 SCCP31.9-4-2SR2-2S
 max-conferences 8 gain -6
 call-park system application
 call-forward pattern .T
 dn-webedit
 time-webedit
 transfer-system full-consult
 transfer-pattern .T
 secondary-dialtone 0
!

The attached document describes in detail what is happening.

I have had this problem for some time. I hope someone can help me.

 

Tank you.

1 Accepted Solution

Accepted Solutions

Vaijanath Sonvane
VIP Alumni
VIP Alumni

Hi,

As per below guide, under SIP Shared-Line (Nonexclusive) section, it noted that when no supplementary-service sip handle-replaces command is configured, SIP shared-line is not supported on CME.  You can try removing this command.

https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/manual/cmeadm/cmebasic.html#concept_4378860158364ABCBAD4EC7F57EAEB23 

 

 

Please rate helpful posts and if applicable mark "Accept as a Solution".
Thanks, Vaijanath S.

View solution in original post

5 Replies 5

Rajan
VIP Alumni
VIP Alumni

Hi

Can you try adding the below:

 

voice register dn 2

shared-line max-calls 4

 

HTH

Rajan

please rate all useful posts by clicking the star below and mark solutions as accepted wherever applicable 

Try playing with the above option and busy trigger per button. That should solve this

Hi.

I tried this, but it didn't work.

Vaijanath Sonvane
VIP Alumni
VIP Alumni

Hi,

As per below guide, under SIP Shared-Line (Nonexclusive) section, it noted that when no supplementary-service sip handle-replaces command is configured, SIP shared-line is not supported on CME.  You can try removing this command.

https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/manual/cmeadm/cmebasic.html#concept_4378860158364ABCBAD4EC7F57EAEB23 

 

 

Please rate helpful posts and if applicable mark "Accept as a Solution".
Thanks, Vaijanath S.

YEH!!
IT WORKS.

Thank you very, very much.