02-12-2018 10:55 PM - edited 03-17-2019 12:11 PM
Hi,
Does anyone encountered the same issue regarding the Hardware Conferencing using SIP IP Phones for CME?
Scenario:
1. Phone A calling Phone B
2. Phone A pressing Conference button and calling Phone C (Phone B is put on hold).
Phone A tried to merge the calls but it is not working.
Note:
1. Hardware Conference bridge status for SCCP is "UP"
2. Debug ephone hw-conference result
*Feb 9 11:16:50.191: Check associated hwconf with callid 3574
*Feb 9 11:16:50.191: Unable to associate a dn/chan with callid 3574
*Feb 9 11:16:56.815: Check associated hwconf with callid 3574
*Feb 9 11:16:56.815: Unable to associate a dn/chan with callid 3574
*Feb 9 11:16:56.815: skinny_hwcfb_check_resource: for Audio
*Feb 9 11:16:56.815: skinny_hwcfbi_find_next_mtpcb Got MTPCB conference type req 52:Audio only as Audio only Pass
*Feb 9 11:16:56.815: skinny_hwcfb_get_adhoc_number:
*Feb 9 11:16:56.815: skinny_hwcfb_get_adhoc_number : Failed to get ad-hoc conference resource
*Feb 9 11:16:59.515: Check associated hwconf with callid 3583
*Feb 9 11:16:59.515: Unable to associate a dn/chan with callid 3583
*Feb 9 11:17:00.683: Check associated hwconf with callid 3574
*Feb 9 11:17:00.683: Unable to associate a dn/chan with callid 3574
*Feb 9 11:17:02.095: Check associated hwconf with callid 3583
*Feb 9 11:17:02.095: Unable to associate a dn/chan with callid 3583
*Feb 9 11:17:02.799: Check associated hwconf with callid 3574
*Feb 9 11:17:02.803: Unable to associate a dn/chan with callid 3574
Best Regards,
Arvie
Solved! Go to Solution.
02-14-2018 07:24 PM
02-14-2018 08:33 PM
02-12-2018 11:05 PM
02-12-2018 11:17 PM
Hello Mohammed,
No.
Cisco 7821 SIP IP Phone configuration.
voice register dn 9
number 28126
allow watch
name EAU - Riche
label EAU - Riche
!
voice register pool 9
busy-trigger-per-button 1
id mac <ip phone mac address>
type 7821
number 1 dn 9
dtmf-relay sip-notify
username 28126 password 28126
codec g711ulaw
Regards,
Arvie
02-12-2018 11:24 PM
02-12-2018 11:27 PM
Hi Mohammed,
I'm using SIP instead of SCCP for my IP Phones so i'm not using ephone-dn.
Regards,
Arvie
02-12-2018 11:36 PM
Hi Mohammed,
Does it mean that Hardware Conference will only work for SCCP and not for SIP IP Phones?
Regards,
Arvie
02-13-2018 01:06 AM
02-13-2018 04:03 AM
Hi Mohammed,
The hardware conference was already added to both voice register global and telephony service but it is still the same.
voice register global
mode cme
source-address <ip address> port 5060
max-dn 50
max-pool 50
load 7821 sip78xx.10-1-1sr1-4
authenticate register
authenticate realm all
timezone 43
time-format 24
hold-alert
tftp-path flash:
file text
create profile sync 0000408060435518
conference hardware
!
telephony-service
sdspfarm units 10
sdspfarm tag 1 d46d.50ab.6660
sdspfarm tag 2 EAUC-MTP
sdspfarm tag 3 conference
conference hardware
max-ephones 58
max-dn 300
ip source-address <ip address> port 2000
max-conferences 8 gain -6
transfer-system full-consult
transfer-pattern .T
create cnf-files version-stamp Jan 01 2002 00:00:00
02-13-2018 06:14 AM
02-14-2018 06:17 PM
Hi,
*** Show SCCP ***
eaucpabx#sh sccp
*Feb 9 11:26:26.619: %SYS-5-CONFIG_I: Configured from console by admin on vty0 (172.22.1.119) all
SCCP Admin State: UP
Gateway Local Interface: GigabitEthernet0/0
IPv4 Address: <ip address>
Port Number: 2000
IP Precedence: 5
User Masked Codec list: None
Call Manager: <ip address>, Port Number: 2000
Priority: 1, Version: 7.0, Identifier: 1
MTP Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: <ip address>, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 2
Reported Max Streams: 40000, Reported Max OOS Streams: 0
Supported Codec: g711ulaw, Maximum Packetization Period: 30
Supported Codec: pass-thru, Maximum Packetization Period: N/A
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
TLS : ENABLED
Conferencing Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: <ip address>, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 1
Reported Max Streams: 160, Reported Max OOS Streams: 0
Supported Codec: g711ulaw, Maximum Packetization Period: 30
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
TLS : ENABLED
SCCP Application Service(s) Statistics:
Profile Identifier: 2, Service Type: MTP
TCP packets rx 210, tx 57
Unsupported pkts rx 0, Unrecognized pkts rx 156
Register tx 1, successful 1, rejected 0, failed 0
Unregister tx 0, successful 0
KeepAlive tx 52, successful 52, failed 0
OpenReceiveChannel rx 0, successful 0, failed 0
CloseReceiveChannel rx 0, successful 0, failed 0
**** Show run ****
sh run
Building configuration...
Current configuration : 34266 bytes
!
!
version 15.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname eaucpabx
!
boot-start-marker
boot system flash:c2900-universalk9-mz.SPA.154-2.T2.bin
boot-end-marker
!
aqm-register-fnf
!
enable password qpalz,901
!
aaa new-model
!
!
!
!
!
!
!
aaa session-id common
clock timezone PHT 8 0
!
!
!
!
!
!
!
!
!
ip domain name eaucpabx.aboitiz.com
ip cef
no ipv6 cef
multilink bundle-name authenticated
!
!
!
!
!
!
!
voice-card 0
dspfarm
dsp services dspfarm
!
!
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip refer
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind media source-interface GigabitEthernet0/0
registrar server
!
!
!
!
voice register pool-type 7821
xml-config maxNumCalls 6
xml-config busyTrigger 6
xml-config custom sshAccess 0
phoneload-support
telnet-support
gsm-support
num-lines 16
description cisco ip phone 7821
reference-pooltype 6921
!
voice register global
mode cme
source-address <ip address> port 5060
max-dn 50
max-pool 50
load 7821 sip78xx.10-1-1sr1-4
authenticate register
authenticate realm all
timezone 43
time-format 24
hold-alert
tftp-path flash:
file text
create profile sync 0000408060435518
conference hardware
!
voice register dn 1
number 28100
allow watch
pickup-call any-group
name EAC - Share
label EAC - Share
!
voice register dn 2
number 28106
allow watch
pickup-call any-group
name EAC - Admin
label EAC - Admin
!
voice register dn 3
number 28102
allow watch
name EAC - Cirilo
label EAC - Cirilo
!
voice register dn 4
number 28105
allow watch
name EAC - HR
label EAC - HR
!
voice register dn 5
number 28103
allow watch
name EAC - Conference
label EAC - Conference
!
voice register dn 6
number 28150
allow watch
name EAC - Grace
label EAC - Grace B
!
voice register dn 7
number 28144
allow watch
name EAC - Riza
label EAC - Riza
!
voice register dn 8
number 28145
allow watch
name EAC - Maria Larisse
label EAC - Maria Larisse
!
voice register dn 9
number 28126
allow watch
name EAC - Riche
label EAC - Riche
!
voice register dn 10
number 28135
allow watch
name EAC - Maintenance
label EAC - Maintenance
!
!
voice register pool 1
busy-trigger-per-button 1
id mac <ip phone mac address>
type 7821
number 1 dn 1
dtmf-relay sip-notify
username 28100 password 28100
codec g711ulaw
!
voice register pool 2
busy-trigger-per-button 1
id mac <ip phone mac address>
type 7821
number 1 dn 2
dtmf-relay sip-notify
username 28106 password 28106
codec g711ulaw
!
voice register pool 3
busy-trigger-per-button 1
id mac <ip phone mac address>
type 8945
number 1 dn 3
dtmf-relay sip-notify
username 28102 password 28102
codec g711ulaw
!
voice register pool 4
busy-trigger-per-button 1
id mac <ip phone mac address>
type 8945
number 1 dn 4
dtmf-relay sip-notify
username 28105 password 28105
codec g711ulaw
!
voice register pool 5
busy-trigger-per-button 1
id mac <ip phone mac address>
type 7821
number 1 dn 5
dtmf-relay sip-notify
username 28103 password 28103
codec g711ulaw
!
voice register pool 6
busy-trigger-per-button 1
id mac <ip phone mac address>
type 7821
number 1 dn 6
dtmf-relay sip-notify
username 28150 password 28150
codec g711ulaw
!
voice register pool 7
busy-trigger-per-button 1
id mac <ip phone mac address>
type 7821
number 1 dn 7
dtmf-relay sip-notify
username 28144 password 28144
codec g711ulaw
!
voice register pool 8
busy-trigger-per-button 1
id mac <ip phone mac address>
type 7821
number 1 dn 8
dtmf-relay sip-notify
username 28145 password 28145
codec g711ulaw
!
voice register pool 9
busy-trigger-per-button 1
id mac <ip phone mac address>
type 7821
number 1 dn 9
dtmf-relay sip-notify
username 28126 password 28126
codec g711ulaw
conference admin
!
voice register pool 10
busy-trigger-per-button 1
id mac <ip phone mac address>
type 7821
number 1 dn 10
dtmf-relay sip-notify
username 28135 password 28135
codec g711ulaw
!
!
license udi pid CISCO2911/K9
hw-module ism 0
!
hw-module pvdm 0/0
!
hw-module pvdm 0/1
!
redundancy
!
!
!
interface Embedded-Service-Engine0/0
no ip address
shutdown
!
interface GigabitEthernet0/0
ip address <ip address>
duplex auto
speed auto
h323-gateway voip interface
h323-gateway voip bind srcaddr <ip address>
!
interface ISM0/0
ip unnumbered GigabitEthernet0/0
service-module ip address <ip address>
!Application: CUE Running on ISM
service-module ip default-gateway <ip address>
!
!
ip forward-protocol nd
!
!
control-plane
!
!
voice-port 0/0/0
supervisory disconnect dualtone mid-call
secondary dialtone
no battery-reversal
no vad
timeouts interdigit 2
timeouts call-disconnect 1
timeouts wait-release 1
connection plar opx immediate 7001
caller-id enable
!
voice-port 0/0/1
supervisory disconnect dualtone mid-call
secondary dialtone
no battery-reversal
no vad
timeouts interdigit 2
timeouts call-disconnect 1
timeouts wait-release 1
connection plar opx immediate 7001
caller-id enable
!
voice-port 0/0/2
supervisory disconnect dualtone mid-call
secondary dialtone
no battery-reversal
no vad
timeouts interdigit 2
timeouts call-disconnect 1
timeouts wait-release 1
connection plar opx immediate 7001
caller-id enable
!
voice-port 0/0/3
supervisory disconnect dualtone mid-call
secondary dialtone
no battery-reversal
no vad
timeouts interdigit 2
timeouts call-disconnect 1
timeouts wait-release 1
connection plar opx immediate 7001
caller-id enable
!
!
!
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
!
mgcp profile default
!
sccp local GigabitEthernet0/0
sccp ccm <ip address> identifier 1 priority 1 version 7.0
sccp
!
sccp ccm group 1
bind interface GigabitEthernet0/0
associate ccm 1 priority 1
associate profile 1 register d46d.50ab.6660
associate profile 2 register EAUC-MTP
keepalive retries 5
switchover method immediate
switchback method immediate
switchback interval 15
!
!
!
dspfarm profile 1 conference
description d46d.50ab.6660
codec g711ulaw
maximum conference-participants 16
maximum sessions 10
associate application SCCP
!
dspfarm profile 2 mtp
description EAUC-MTP
codec g711ulaw
codec pass-through
maximum sessions software 20000
associate application SCCP
!
!
sip-ua
registrar ipv4:<ip address> expires 3600
sip-server ipv4:<ip address>
!
!
!
gatekeeper
shutdown
!
!
telephony-service
sdspfarm units 10
sdspfarm tag 1 d46d.50ab.6660
sdspfarm tag 2 EAUC-MTP
sdspfarm tag 3 conference
conference hardware
max-ephones 58
max-dn 300
ip source-address <ip address> port 2000
max-conferences 8 gain -6
transfer-system full-consult
transfer-pattern .T
create cnf-files version-stamp Jan 01 2002 00:00:00
!
end
02-14-2018 07:24 PM
02-14-2018 07:25 PM
02-14-2018 07:49 PM
Hi Nipun Singh Raghav,
Do you mean that I need to create ephone-dn for my SIP IP Phone even though i'm using voice register pool for my SIP IP Phones?
Regards,
Arvie
02-14-2018 08:33 PM
02-20-2018 04:53 PM
Hi Nipun Singh Raghav,
I’ve already tested it with 5 participants and so far it is working. I’ll try to create up to 10 participants and see if it will work also. Thank you for the information.
Arvie
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