06-10-2017 04:33 AM - edited 03-18-2019 12:16 PM
Hi all,
When call coming from PSTN, CME registered SIP phone answer the call, but PSTN side phone still keeps ringing. CME is registered to SIP provider.
SIP phone--CME---SIP Provider---PSTN---phone
Here is CME configuration:
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
no supplementary-service sip handle-replaces
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
registrar server
!
voice register pool-type 8865
phoneload-support
transport tcp
description CISCO SIP PHONE 8865
reference-pooltype 9971
!
voice register global
mode cme
source-address 192.168.10.1 port 5060
max-dn 10
max-pool 35
authenticate register
timezone 22
tftp-path flash:
create profile sync 0005778351327525
camera
video
auto-register
!
voice register dn 1
number 1001
!
voice register dn 2
number 1002
!
voice register dn 3
number 1003
!
voice register dn 4
number 1004
!
voice register dn 5
number 1005
!
voice register pool 1
busy-trigger-per-button 2
id mac 9C57.ADD2.0EBF
type 7841
number 1 dn 5
username 1001 password 1001
codec g711ulaw
!
voice register pool 2
busy-trigger-per-button 2
id mac 9C57.ADD2.0E78
type 7841
number 1 dn 2
username 1002 password 1002
codec g711ulaw
!
voice register pool 3
busy-trigger-per-button 2
id mac 9C57.ADD2.0EB1
type 7841
number 1 dn 3
username 1003 password 1003
codec g711ulaw
!
voice register pool 4
busy-trigger-per-button 2
id mac 9C57.ADD2.0E9B
type 7841
number 1 dn 4
username 1004 password 1004
codec g711ulaw
!
voice register pool 5
id mac 94D4.690C.D981
type 8865
number 1 dn 1
username 1005 password 1005
codec g711ulaw
camera
video
voice translation-rule 1
rule 1 /^....$/ /12027981446/
!
voice translation-rule 2
rule 3 /.*/ /1005/
voice translation-profile INBOUND
translate called 2
!
voice translation-profile OUTBOUND
translate calling 1
dial-peer voice 22 voip
description Mobile
preference 1
destination-pattern 088.......
session protocol sipv2
session target ipv4:192.168.10.22
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 33 voip
description Mobile
preference 1
destination-pattern 077.......
session protocol sipv2
session target ipv4:192.168.10.22
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 1 voip
description International
translation-profile outgoing OUTBOUND
preference 1
destination-pattern .T
session protocol sipv2
session target dns:*******
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 2 voip
description Mobile
translation-profile outgoing OUTBOUND
preference 2
destination-pattern 088.......
session protocol sipv2
session target dns:*******
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 3 voip
description Mobile
translation-profile outgoing OUTBOUND
preference 2
destination-pattern 077.......
session protocol sipv2
session target dns:*******
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 100 voip
description Incoming calls
translation-profile incoming INBOUND
incoming called-number .
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
sip-ua
credentials username ******* password 7 ******* realm *******
authentication username ******* password 7 *******
nat symmetric role active
nat symmetric check-media-src
registrar dns:*******:5060 expires 3600
sip-server dns:*******:5060
connection-reuse
host-registrar
telephony-service
no auto-reg-ephone
max-ephones 35
max-dn 35
ip source-address 192.168.10.1 port 2000
cnf-file location flash:
max-conferences 8 gain -6
transfer-system full-consult
transfer-pattern .T
create cnf-files version-stamp Jan 01 2002 00:00:00
06-11-2017 07:21 AM
Please post a "debug ccsip messages" output for the problematic call.
06-12-2017 01:35 AM
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